On Thu, 2009-02-19 at 15:22 +1300, Michael wrote:
This is everything that is wrong with Open Source - no body wants to pay for
anything
Your statement is not correct!
Well atleast half of it. You should have said:
-no body wants to pay for anything-
This has absolutely nothing to do
Hi,
Few subjects cause as many arguments as which SIP client works best?
on IRC #asterisk, voip forums, and probably the -users mailing list. I
have tried most of the SIP clients available in the last 5 years, both
with Asterisk and other platforms such as OnSIP.com, IConnectHere.com,
ZipDX.com
This has absolutely nothing to do with the fact that something is
opensource. The fact that the source is open has nothing todo with its
pricetag. Sometimes opensource products are more expensive then closed
source products.
If you want support/maintenance/dedicated_features/you-name-it
Big companies, especially those with major computing systems use paid
software
because they want a vendor they can hold responsible for it.
As for OSS and FOSS, it is majorly used by the sort of businesses and
individuals who call me (and other IT pros) up and talk the talk, but they
2009/2/18 Asterisk Asterisk nt_aster...@yahoo.com
Thanks for the feedback. I did some research and it looks like you were
calling over international lines. It also appears that there was high than
average static on the line, which is not normal for my system. It's true
that I threw my
Darren Wiebe wrote:
Pretty cool. I'm almost offended though as I'm not usually guessed as a
female of the species. :)
I am a male and was detected as a male, so I'm feeling a bit left out. :-p
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2009/2/19 Johansson Olle E o...@edvina.net
19 feb 2009 kl. 07.47 skrev Olivier:
Hi,
I've been asked sometimes to tailor call history features embeded in
SIP hardphones.
For example, a cutomer wanted internal call to be taken out.
Another wanted calls to sorted according specific
I think in this case when 5k call are involved i think all the difficulty of
the project is to split the load on different parts of the system. In my
case i would do it like that:
Phones ---Opensips (Double server with heartbeat and in different places)
|
Hi
Im using a sip phone SPA921, and the one that calls me can hear me but I cant
hear them, when I make the call I can hear them.
Im running asterisk 1.6 behind a firewall, I have port 1-2 for rtp and
5060 for sip forward to my asterisk.
Any tips?
Regards
/ralf
Hi all,
I am using Asterisk-1.6.0.1 version and done below configuration
for the extensions.conf.
exten = x-lite1,hint,SIP/x-lite1
The x-lite1 phone is configured in sip.conf.
Now I(Application) gets subscribe for x-lite1 for the events.
But when I make a call from x-lite1 to
Hi all, I'm searching for a way to inform my Cisco CME that a number on
Asterisk server is busy.
I have a SIP trunk between Cisco and Asterisk and some Cisco ip phone
have a speed dial with a number registered on Asterisk.
How can I exchange busy information between two PBX?
Thanks Enrico.
Michael wrote:
This has absolutely nothing to do with the fact that something is
opensource. The fact that the source is open has nothing todo with its
pricetag. Sometimes opensource products are more expensive then closed
source products.
If you want
Hello all,
I'm trying to prevent answering a channel when a queue is either full or has
no members. It seems I'm forced to answer a call before I call Queue() or
else the audio is in the early media (which is unacceptable because of the
short duration of early media on ISDN).
Is there any
Johansson Olle E schrieb:
19 feb 2009 kl. 00.08 skrev Klaus Darilion:
Hi!
I have some problems understanding the concept of REFER in Asterisk
1.4.23.
I have the following scenario:
Incoming SIP call (incoming leg) from a SIP trunk into Asterisk
(handled
in context fromTrunk),
Hi,
I have a snom 360 connected to two asterisk servers(both 1.6.0.5), via
two identities. Each asterisk server runs a queue and snom is a
member of queue in both servers. Currently when snom is receiving call
from one asterisk server, it can still receive a call from the other
asterisk, because
\
No... there's no shortage of work that needs doing. But there's a
definite shortage of money to pay those to do it -- hence the massive,
worldwide layoffs. Your little corner may not be affected, but to
discount basic economics because you don't see it? Well... that's
incredibly
I am looking for someone that could share their code for this function:
Outgoing call - macro that checks if line is (not human) or machine,
fax, busy, subscriber problem and other fault tones - if human connect
to agent else hangup and write status to cdr.
Need help with this!
Regards
You can know if the queue is full before issuing the answer() or the queue()
command, so you can avoid answering at all.
l.
2009/2/19 Alex Hermann a...@speakup.nl
Hello all,
I'm trying to prevent answering a channel when a queue is either full or
has
no members. It seems I'm forced to
One simple thing that comes to my mind is to have the SNOM connected to only
one server, and send calls to from the queue on the second server to the
first server, so that you can enforce a acall limit.
l.
2009/2/19 Rajkumar S rajkum...@gmail.com
Hi,
I have a snom 360 connected to two
Hi,
Can somebody please shed some light on how to use the
QUEUE_VARIABLES() function?
The built-in help says
---cut---
Return Queue information in variables
[Description]
Makes the following queue variables available.
QUEUEMAX maxmimum number of calls allowed
QUEUESTRATEGY the strategy of the
Rajkumar S schrieb:
I have a snom 360 connected to two asterisk servers(both 1.6.0.5), via
two identities. Each asterisk server runs a queue and snom is a
member of queue in both servers. Currently when snom is receiving call
from one asterisk server, it can still receive a call from the
On Wed, 11 Feb 2009, Philipp Kempgen wrote:
Jeff LaCoursiere schrieb:
Working on some niche requests from one of my hotel clients. asterisk
1.4.20-1 on CentOS, Polycom 501s.
The first request is for the Polycom's screen to show the CID of the
inbound caller when a call pick is executed,
Alex Hermann schrieb:
I'm trying to prevent answering a channel when a queue is either full or has
no members. It seems I'm forced to answer a call before I call Queue() or
else the audio is in the early media (which is unacceptable because of the
short duration of early media on ISDN).
On Thu, Feb 19, 2009 at 8:28 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
Easy solution: Disable call waiting on the phone.
But asterisk will attempt a call since it's status is idle, and will
generate events which will confuse ADM I am using to display a url
for call.
Advanced
On Thu, 2009-02-19 at 15:54 +0100, Philipp Kempgen wrote:
Can somebody please shed some light on how to use the
QUEUE_VARIABLES() function?
I'll try...
How is it supposed to be called?
Set(void=${QUEUE_VARIABLES(techsupport)}); ?
That's how it is supposed to be called, but it obviously
On Thursday 19 February 2009, Lenz Emilitri wrote:
You can know if the queue is full before issuing the answer() or the
queue() command, so you can avoid answering at all.
I don't think the 'maxlen' option is available from the dialplan. Further, the
QUEUE_MEMBER_COUNT function doesn't work
Rajkumar S schrieb:
On Thu, Feb 19, 2009 at 8:28 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
Easy solution: Disable call waiting on the phone.
But asterisk will attempt a call since it's status is idle,
Unfortunately yes.
and will
generate events which will confuse ADM I am
Max Alex schrieb:
I have configured the phpagi application for counting the duration of call,
The call is originated from the script and after hangup the call the
duration and status will be stored.
This functionality and php script is working fine with deadagi application
with asterisk 1.4.
there is an application amd() but it is not perfect.
i get better result using the dial timeout if it rings 3 times hangup or a
machine will answer :-)
you always can call again later.
David
2009/2/19 Marcus Kvarsell mar...@bizint.se
I am looking for someone that could share their code for
David fire wrote:
there is an application amd() but it is not perfect.
i get better result using the dial timeout if it rings 3 times hangup
or a machine will answer :-)
you always can call again later.
David
its stuff like this which makes telemarketers hated - you're going to
hang up and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Olivier a écrit :
I've got a customer (a University) who should be interested in advanced
call cost control techniques.
Beside limiting dialing, reading AOC-D messages would be help to keep
those costs down.
Are these AOC-D data reliable for
This problem is only going to get worse as the so-called 'recession' bites...
fellow I.T. professionals - get used to your clients trying to weasel free
service out of you. Everything I am hearing from fellow I.T. people is that
there is no shortage of 'work' but a lot of clients are resisting
Jon Pounder escribió:
David fire wrote:
there is an application amd() but it is not perfect.
i get better result using the dial timeout if it rings 3 times hangup
or a machine will answer :-)
you always can call again later.
David
I think that the AMD application accuracy depends
On 18/02/2009 9:28 p.m., Rajkumar S wrote:
Hi,
Russell's blog[1] is down and there are not much information about
this any where else. Any one with more information about res_ais and
how it is used?
His blog is back up, but google cache, Daily Asterisk News etc
--
Kind Regards,
Matt
Please don't cross-post.
Max Alex schrieb:
I am using asterisk 1.4.19,
I have setup the dialplans to get the incoming call and that will be sent to
another context by local channel,
In another context i have setup the ring group, that portion is working
fine.
I have noticed that when i have
Gordon Henderson schrieb:
On Tue, 17 Feb 2009, Gordon Henderson wrote:
It looks like something has changed in the HPET kernel code in 2.6.28
(maybe .27 too) that's stopped ztdummy.c compiling (in 1.2 and 1.4
versions of zapata) A kernel structure member has been renamed with some
crypic
According to the FTC, you must let the phone ring for at least 30 seconds
before hanging up.
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Hi Tony,
There are quite a few differences between the two modes.
1) connections
Component connections use a different TCP port than the regular ports
(5222 or 5223) use for client connections. For example, the default
port on jabberd2 for component connections is 5347. Component
connections
On Feb 18, 2009, at 10:47 PM, Olivier wrote:
Hi,
I've been asked sometimes to tailor call history features embeded in
SIP hardphones.
For example, a cutomer wanted internal call to be taken out.
Another wanted calls to sorted according specific criteria.
1. Have you identified a phone
Date: Wed, 18 Feb 2009 09:50:18 -0800
From: bilmar...@yahoo.com
Subject: Re: Credit Card processing machines
To: asterisk-users@lists.digium.com
CC: listas_quij...@hotmail.com
And is there a bank accept to give such kind of communication?
The user was able to dial his card number
Why not Asterisk?
And if need to use RS232, then ethernet is not possible? So how u will use AGI
with RS232?
Regards
Bilal
--- On Thu, 2/19/09, Edwin Quijada listas_quij...@hotmail.com wrote:
From: Edwin Quijada listas_quij...@hotmail.com
Subject: RE: Credit Card processing machines
To:
On Thu, 19 Feb 2009, bilal ghayyad wrote:
So how u will use AGI with RS232?
While an AGI can talk to /dev/ttySx, it would make more sense to write a
daemon to manage the conversations and then your AGIs could talk to the
daemon over TCP, shared memory, pipes, two cans and some string...
Hello,
I have several customers describing something like annoying silence
suppression. So I did some tests and I can confirm. After disabling
echo cancellation in zapata.conf the silence suppression effect is
gone, but there is a little echo of course.
I do not have this problem on other boxes
Date: Thu, 19 Feb 2009 11:25:50 -0800
From: bilmar...@yahoo.com
Subject: RE: Credit Card processing machines
To: asterisk-users@lists.digium.com; listas_quij...@hotmail.com
Why not Asterisk?
And if need to use RS232, then ethernet is not possible? So how u will use
AGI with RS232?
You could use the XML browser on the cisco 79xx series.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Chamberlain
Sent: Thursday, February 19, 2009 2:18 PM
To: Asterisk Users Mailing List -
IVR Number :17275691533
When I try it from xlite configuring my provider directly, it works
perfectly.
When I try to dial out from dialer , it doesnt work.
[sip8]
type=peer
username=user
fromuser=user
authuser=user
secret=password
host=8.14.146.111
nat=no
canreinvite=yes
insecure=very
Is your soft phone also using rfc2833 for DTMF mode?
--Tim
- David @ULC wrote:
IVR Number :17275691533
When I try it from xlite configuring my provider directly, it works
perfectly.
When I try to dial out from dialer , it doesnt work.
[sip8]
type=peer
username=user
How can I detect how many ring a call to hangup?
Where I can find info about AMD?
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo
On 19 Feb 2009, at 20:48, David @ULC wrote:
IVR Number :17275691533
When I try it from xlite configuring my provider directly, it works
perfectly.
When I try to dial out from dialer , it doesnt work.
[sip8]
type=peer
username=user
fromuser=user
authuser=user
secret=password
Loic Didelot wrote:
I would really like to use the hardware echo canceller I bought, is
there any way to tune it or change a few things?
This is probably a known issue with the firmware for that echo
canceller; a new version is available, please contact Digium Support for
assistance obtaining
--- (12 headers 0 lines) ---
Sending to 192.168.0.50 : 12714 (NAT)
Transmitting (NAT) to 192.168.0.50:12714:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.50:12714
;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714
From: cc106sip:cc...@192.168.0.2
Don't most?
From: Nhadie nha...@gmail.com
To: Asterisk-users@lists.digium.com
Sent: Wednesday, February 18, 2009 6:19:24 AM
Subject: [asterisk-users] US DID
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
Hello all,
I have two machines I'm connecting with TDMOE (dahdi dynamic spans) and
I have a question about timing parameters. By my understanding one
machine should be the source of the timing and the other a slave of that
timing.
So on machine A I have the following in system.conf:
Not entirely. I've been trying for two years to get someone to work with
my small Linux system. One guy never had time to come. I finally got
someone out who was going to charge either $125 or $175 per hour (USD)
depending on whether he decided it was a computer problem or a network
You can probably use combo of NVLineDetect, NVGenderDetect, and AMD
(NVMachineDetect).
From: Edwin Quijada listas_quij...@hotmail.com
To: Asterisk Asterisk asterisk-users@lists.digium.com
Sent: Thursday, February 19, 2009 12:55:05 PM
Subject: Re:
You might also check with www.star2star.com (Star2Star Communications). We did
a call park, pickup, and transfer module with similar functionality. Integrates
very nicely.
Justin Newman
nt_jnewman at yahoo.com
From: Jeff LaCoursiere j...@jeff.net
To:
Try canreinvite=update instead along with directrtpsetup=yes.
On Wed, Feb 18, 2009 at 9:58 AM, Arturo Díaz Almagro
arturo.diaz.alma...@gmail.com wrote:
Hello,
I have a working system based on asterisk 1.4.23.1 and I want RTP going
end-to-end but not using canreinvite because it creates
We have a BLF module that maintains device state across Asterisk servers.
Contact me off the list if interested.
Justin Newman
nt_jnewman at yahoo.com
From: Lenz Emilitri lenz.lo...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi!
Advanced solution: Use local channels as queue members and Custom
hints. You could build a mechanism (outside of Asterisk) to sync
the states of your Custom hints between both servers.
I am already using local channels and will explore hints. I have not
used it till now, any hints
You sure you don't have a pony tail? :) Hehe.
It happens to the best of us. Hopefully after my fine tuning it will happen to
less of us!
From: Darren Wiebe dar...@aleph-com.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
NVLineDetect, NVGenderDetect what is that?
amd info voip-info.org or asterisk.org support asterisk book.
i bougth one to support the cause!!!
David
2009/2/19 Asterisk Asterisk nt_aster...@yahoo.com
You can probably use combo of NVLineDetect, NVGenderDetect, and AMD
(NVMachineDetect).
Dear All,
I would like to ask please if someone has a AGI script that select a value
from a database and dial this value as a destination number
Regards
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Sure,
what did you need exactly ?
On Thu, Feb 19, 2009 at 6:57 PM, michel freiha mich...@gmail.com wrote:
Dear All,
I would like to ask please if someone has a AGI script that select a value
from a database and dial this value as a destination number
Regards
Dear Sir,
I need the followingA customer will dial a specific number like 112,this
will fire a php AGI script...The AGI script will connect to the database and
see if this number (112) is registered in a specific table...If yes, AGI
script will return to asterisk a destination number
On Fri, 20 Feb 2009, michel freiha wrote:
I need the followingA customer will dial a specific number like 112,this
will fire a php AGI script...The AGI script will connect to the database and
see if this number (112) is registered in a specific table...If yes, AGI
script will return to
It got my gender correct the two times I tested, even with the TV loud in
the background.
BTW, I love the beep.
On Thu, Feb 19, 2009 at 5:54 PM, Asterisk Asterisk nt_aster...@yahoo.comwrote:
You sure you don't have a pony tail? :) Hehe.
It happens to the best of us. Hopefully after my fine
On Thu, Feb 19, 2009 at 5:04 PM, Michael mich...@networkstuff.co.nz wrote:
Not entirely. I've been trying for two years to get someone to work with
my small Linux system. One guy never had time to come. I finally got
someone out who was going to charge either $125 or $175 per hour (USD)
I also think you should check the economic stimulus package. There are
billions of dollars allocated to ISPs. It could be a windfall.
Yoohoo! Let's print some more money. I don't think that's been tried
before
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On Thu, Feb 19, 2009 at 5:00 PM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Hello all,
I have two machines I'm connecting with TDMOE (dahdi dynamic spans) and
I have a question about timing parameters. By my understanding one
machine should be the source of the timing and
On Fri, 20 Feb 2009, Michael wrote:
I also think you should check the economic stimulus package. There are
billions of dollars allocated to ISPs. It could be a windfall.
Yoohoo! Let's print some more money. I don't think that's been tried
before
You must of course know all about
did you get your Samba issue fixed?
Yes, I finally figured it out this morning. I had the domain set to match that
of the web site it was serving, which was on the other side of a router, so not
even on the local subnet. The DNS queries from that got Samba all messed up.
Ugh!
Yes, I would
You must of course know all about how things are working here in the
states while you sit in NZ. The money is not being printed. It is being
I am well informed thanks to the global reach of such excellent purveyours of
quality news and information such as Fox and CNN.
raised and accounted
On Thu, Feb 19, 2009 at 7:46 PM, Michael mich...@networkstuff.co.nz wrote:
I also think you should check the economic stimulus package. There are
billions of dollars allocated to ISPs. It could be a windfall.
Yoohoo! Let's print some more money. I don't think that's been tried
On Fri, 2009-02-20 at 13:46 +1300, Michael wrote:
I also think you should check the economic stimulus package. There are
billions of dollars allocated to ISPs. It could be a windfall.
Yoohoo! Let's print some more money. I don't think that's been tried
before
Of course it has!
On Thu, Feb 19, 2009 at 8:02 PM, Michael mich...@networkstuff.co.nz wrote:
You must of course know all about how things are working here in the
states while you sit in NZ. The money is not being printed. It is being
I am well informed thanks to the global reach of such excellent
Hey I am against any Federal bailouts (wrote in Ron Paul and donated to his
campaign for presidency as well as being an active member of the Campaign
for Liberty http://campaignforliberty.com) , but they are a reality.
Shame Ron Paul was shafted of fair media attention. He would be an
Jeff LaCoursiere wrote:
On Fri, 20 Feb 2009, Michael wrote:
I also think you should check the economic stimulus package. There are
billions of dollars allocated to ISPs. It could be a windfall.
Yoohoo! Let's print some more money. I don't think that's been tried
before
Money is borrowed at the Prime Rate which is between 0% and .25% but thus
far, China and Japan have been lending the largest portions as well as
other foreign countries. I am not saying this is any better but you should
get your facts straight.
I am well aware of this. I am also well aware
On Thu, 19 Feb 2009, John Novack wrote:
This will take a very long time to work itself out, and folks better get
used to some very lean times.
Be thankful you have a marketable skill.
Not that it seems to be getting me any interviews :)
j
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Is this the asterisk users list? or some political list? or maybe i
dunno, i am confused
On Thu, Feb 19, 2009 at 8:19 PM, Michael mich...@networkstuff.co.nz wrote:
Money is borrowed at the Prime Rate which is between 0% and .25% but thus
far, China and Japan have been lending the
i dont see the connection with AGI pdf book
On Thu, Feb 19, 2009 at 8:28 PM, Emmanuel Bruno tipas...@gmail.com wrote:
Is this the asterisk users list? or some political list? or maybe i
dunno, i am confused
On Thu, Feb 19, 2009 at 8:19 PM, Michael mich...@networkstuff.co.nzwrote:
On Thu, 19 Feb 2009, Philipp Kempgen wrote:
Rajkumar S schrieb:
and will generate events which will confuse ADM I am using to display a
url for call.
ADM?
Asterisk Desktop Manager. http://adm.hamnett.org/
core show function DEVICE_STATE (on 1.6) is a good start.
Thanks.
raj
At 04:23 PM 2/19/2009, you wrote:
It got my gender correct the two times I tested, even with the TV
loud in the background.
It got me wrong twice, but so do about 30% of the people who call.
Ira
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On Fri, 20 Feb 2009 14:28:23 you wrote:
Is this the asterisk users list? or some political list? or maybe i
dunno, i am confused
Sorry. I'll make that my last post on the subject.
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Broadvox
Nhadie wrote:
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
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i have in perl AGI script if u want then let me know ...?
On Fri, Feb 20, 2009 at 4:59 AM, Steve Edwards asterisk@sedwards.comwrote:
On Fri, 20 Feb 2009, michel freiha wrote:
I need the followingA customer will dial a specific number like
112,this
will fire a php AGI script...The
2009/2/19 Steve Edwards asterisk@sedwards.com
On Fri, 20 Feb 2009, michel freiha wrote:
I need the followingA customer will dial a specific number like
112,this
will fire a php AGI script...The AGI script will connect to the database
and
see if this number (112) is registered in
Setup is:
Asterisk ---NAT-- SIP Proxy
I have following entry for SIP Proxy in sip.conf
[Proxy]
type=peer
host=Static IP (NAT Firewalls public IP)
username=
secret=x
nat=yes
canreinvite=no
qualify=yes
Proxy sends a call and I get this error
Found no matching
Lenz Emilitri schrieb:
I think this is by design - each time the Dial() is performed, SIP
headers are reset.
No.
SIPAddHeader adds global channel variables to the incoming channel. Dial
copies the global variables to the outgoing channel. If the outgoing
channel is a SIP channel, the
89 matches
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