I still think we need a SIP_CAUSE channel variable. :-)
Then we need to start working on aggregation rules, like what if one
IAX channel answers and one SIP channel is busy?
For SIP-only calls, we need to add a lot of code from proxy rules for
call forking and response aggregation. It's
Hello.
Does someone know what order field means in followme.conf? The Doku says:
number= number to call[2nd #[3rd #]] [, timeout value in seconds [,
order in follow-me] ]
So an example would be:
number= 123124125,10,?
It would be nice if someone could enlighten me.
cheers
t.
Johansson Olle E schrieb:
I still think we need a SIP_CAUSE channel variable. :-)
Then we need to start working on aggregation rules, like what if one
IAX channel answers and one SIP channel is busy?
For SIP-only calls, we need to add a lot of code from proxy rules for
call forking
Hi!
If a user has recorded a new voicemail message (e.g. unavailable
message) then it is prompted with 3 choices.
1. accept recording
2. listen to the recorded message
3. rerecord the message
Isn't it possible to cancel the recording?
thanks
klaus
Hello,
I am missing any description of zaptel/DAHDI alarms. The TE200 series
user manual contains only a description of LEDs states. These alarms
states are visible in zttool/dahditool or in astersick CLI (zap show
status) and I wonder what is the real meaning of these alarms for E1
channel.
Hi Thomas Stein
this is the syntax of follow me
exten = s,5,Macro(stdexten-followme,${ARG1},${ARG2})
On Mon, Jan 19, 2009 at 4:38 PM, Thomas Stein
thomas.st...@knowledgetools.de wrote:
Hello.
Does someone know what order field means in followme.conf? The Doku says:
number= number to
Klaus Darilion schrieb:
If a user has recorded a new voicemail message (e.g. unavailable
message) then it is prompted with 3 choices.
1. accept recording
2. listen to the recorded message
3. rerecord the message
Isn't it possible to cancel the recording?
You could hang up.
But users
Hi,
Not sure where to submit this to so I'll try here. Below is the toneset for
Iceland. Hopefully this can be added into the asterisk package.
[is]
description = Iceland
ringcadence = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/5000
congestion = 425+250/250,0/250
callwaiting =
Dear All,
I have the following CPU info on my asterisk server:
Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST
2008 i686 i686 i386 GNU/Linux
I need to install G729 on the asterisk server just to pass through and not
for encoding...Which G729 package do you advice me
1)--Download G729 modules compatible with cpu model and asterisk version
http://asterisk.hosting.lv/
2)--Change module
rename to codec_g729.so
copy to /usr/lib/asterisk/modules
set permission 755
3)--
restart asterisk
coonect to asterisk and type 'show translation'
please attach:
cat /proc/cpuinfo
2009/1/19 michel freiha mich...@gmail.com
Dear All,
I have the following CPU info on my asterisk server:
Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST
2008 i686 i686 i386 GNU/Linux
I need to install G729 on the asterisk
if u have problem :
4)--Disable selinux
Go to /etc/selinux/ and type (vim config)
comment All lines
reboot your linux
From: morteza kashani kasha...@yahoo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk does pass thru out of the box, there is nothing to install.
in your sip.conf
just add the following:
disallow=all
allow=g729
this will force the peer to use g729 and the end points will take care of the
codec assuming both end points support g729 to begin with.
-jon
-
hi
just for pass through you dont need any codec...
2009/1/19 michel freiha mich...@gmail.com
Dear All,
I have the following CPU info on my asterisk server:
Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST
2008 i686 i686 i386 GNU/Linux
I need to install G729 on
On 1/19/2009 12:03, michel freiha wrote:
Dear All,
I have the following CPU info on my asterisk server:
Linux switch1.domain.net http://switch1.domain.net 2.6.18-92.1.22.el5
#1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux
I need to install G729 on the asterisk server just to
Hi All;
Anyone knows an IAX IP Phone works fine and tested?
Does polycom support IAX IP Phone?
Regards
Bilal
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asterisk-users mailing list
To UNSUBSCRIBE or update
Johansson Olle E schrieb:
Even if I think there's only one protocol for the future
Which is? :-) SIP? Maybe XMPP?
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH -
Hi
When we ned to call 112 (emergency number) we need to add 0379 before 112 and
464 after for it to work, how do I do that In my dialplan?
The caller should only dial 112 on the phone.
Regards
/ralf
Ralf Träskman, IT
AdLibris AB, Odengatan
exten = 112,1,Dial(SIP/Provider/0379464${EXTEN})
bye
2009/1/19 Ralf Träskman r...@adlibris.com
Hi
When we ned to call 112 (emergency number) we need to add 0379 before 112
and 464 after for it to work, how do I do that In my dialplan?
The caller should only dial 112 on the phone.
sorry try with:
exten = 112,1,Dial(SIP/Provider/0379${EXTEN}464)
2009/1/19 Daniel Ortiz zate...@gmail.com
exten = 112,1,Dial(SIP/Provider/0379464${EXTEN})
bye
2009/1/19 Ralf Träskman r...@adlibris.com
Hi
When we ned to call 112 (emergency number) we need to add 0379 before 112
Hi
Thanks
/ralf
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Ortiz
Sent: den 19 januari 2009 14:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] adding numbers in dialplan
sorry
How to add SipAddHeader in outgoing call file.
I am implementing a Callback scenario, in which a user makes a call to
Local Access Number. The system have to callback to the user. During
callback a call file is generated. All I want, is to add
SipAddHeader(pchargingvector,val) in outgoing Invite.
Hi everyone,
In one of my contexts I run h priority in which I need to change the
CDR(dst) value. But it doesn't work and in the CDR dst field is recorded as
h.
Context abc {
111 = {
...
...
...
};
h = {
Set(CDR(dst)='111');
NoOp(${CDR(dst)});
Hangup();
};
};
Can anybody give me an idea how
I have just got a Cisco 7941G and am experiencing the exact same
problem (phone is requesting .tlv file from TFTP server and never asks
for .cnf.xml file). The phone originally had SCCP on it, but I
downloaded and flashed with the latest Cisco SIP image (8.4(3)
released 2009-01-13). In
Dear Sir,
kindly find below my CPU info...I just need which package should i install
[r...@switch1 modules]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 2
model name : Intel(R) Xeon(TM) CPU 3.20GHz
stepping: 5
cpu MHz
First, thanks for your help
Ok, i going to do a script and call ot with only one 'System' (cf Gordon
Henderson) and take a look to 'incron' (T Cohen)
Just need some explanations:
1) If the call file 'failed', an 'exitstatus' is happendGood
How to check/get these $ and put in in an * $ ?
19 jan 2009 kl. 11.10 skrev Philipp Kempgen:
Johansson Olle E schrieb:
I still think we need a SIP_CAUSE channel variable. :-)
Then we need to start working on aggregation rules, like what if one
IAX channel answers and one SIP channel is busy?
For SIP-only calls, we need to add a lot
Hi All
I got Asterisk to run on Solaris however I do need it to run in
realtime mode I.e. with the res_mysql file.
Did anyone succeed in this ?
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
bilal ghayyad wrote:
Hi All;
Anyone knows an IAX IP Phone works fine and tested?
Does polycom support IAX IP Phone?
Regards
Bilal
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
On Mon, 2009-01-19 at 08:45 -0500, Zeeshan Zakaria wrote:
Hi everyone,
In one of my contexts I run h priority in which I need to change the
CDR(dst) value. But it doesn't work and in the CDR dst field is
recorded as h.
Context abc {
111 = {
...
...
...
};
h = {
- sean darcy seandar...@gmail.com wrote:
OK. Calmer now. If fact a 410 would have the same problem.
I'll make the fix on our machines. Should I file a bug, or does the
169154 commit already fix it?
sean
The issues has been corrected in trunk and the 1.6.1 branch.
Sicne we
Dear David;
At what price u get it?
Did u test it with IAX and SIP? Are u sure it is good? As really I did not deal
with chinese phone until now and I found it fine.
Regards
Bilal
--- On Mon, 1/19/09, David da...@linuxcrazy.com wrote:
From: David da...@linuxcrazy.com
Subject: Re:
The reason why I introduced h priority here is that I needed to get the
variable CDR(duration) for DeadAGI script which I am also running in h
priority. Without h priority, I was getting correct CDR(dst) value but not
correct CDR(duration) value even if I tried to run DeadAGI after Hangup().
Is it possible for asterisk to send sms through a GSM gateway, tor example
the Portech MV-37X?
If yes, any examples of configurations would be really apreciated.
On Tue, Oct 14, 2008 at 11:13 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
The most flexible way but will require a bit
Johansson Olle E schrieb:
19 jan 2009 kl. 11.10 skrev Philipp Kempgen:
Johansson Olle E schrieb:
I still think we need a SIP_CAUSE channel variable. :-)
Then we need to start working on aggregation rules, like what if one
IAX channel answers and one SIP channel is busy?
For SIP-only
On Mon, 2009-01-19 at 11:51 +, Örn Arnarson wrote:
Not sure where to submit this to so I'll try here. Below is the
toneset for Iceland. Hopefully this can be added into the asterisk
package.
Could you please add it to the request tracker at
http://bugs.digium.com, so that it doesn't get
Hi,
Is anyone using Fring as a SIP client to an Asterisk server ?
A prospective customer of mine is asking to integrate its iphones with an
Asterisk server and after googling, I still have some unanswered questions :
1. Which codecs are available when calling from fring ?
2. Is it easy and
On Mon, 2009-01-19 at 11:10 +0100, Lukas Rypl wrote:
I am missing any description of zaptel/DAHDI alarms. The TE200 series
user manual contains only a description of LEDs states. These alarms
states are visible in zttool/dahditool or in astersick CLI (zap show
status) and I wonder what is the
On Mon, 19 Jan 2009, bilal ghayyad wrote:
Hi All;
Anyone knows an IAX IP Phone works fine and tested?
Does polycom support IAX IP Phone?
Regards
Bilal
How about IAX2 adapter from digium?
I've been uing it and it works very well.
--
#Joseph
GPG KeyID: ED0E1FB7
On Mon, 19 Jan 2009, Pascal Bruno wrote:
Is it possible for asterisk to send sms through a GSM gateway, tor example
the Portech MV-37X?
If yes, any examples of configurations would be really apreciated.
AIUI, the Portechs can recieve TXTs and you can see them via their Web
interface.. I
Hi All
I'm having some serious kernel panic while using digium cards.
It may be related to IRQ shared.
Can this cause a lot of drop call and bad voice quality ?
Do you guys know if there is a way I can assign one IRQ for each digium card
?
Thanks a lot.
Here is the output of
Hi everyone,
I googled this followed the instructions, but it hasn't work for me yet.
I have universal setting in SIPDefault.cnf and phone specific settings in
SIPXX.cnf. But it doesn't get registered.
I need to register it on two different asterisk boxes. So my
SIPXX.cnf looks
on kernel boot parameters do it:
acpi=off
Regards,
Luis Morales
On Mon, Jan 19, 2009 at 12:25 PM, Plugworld plugwo...@micnes.com wrote:
Hi All
I'm having some serious kernel panic while using digium cards.
It may be related to IRQ shared.
Can this cause a lot of drop call and bad
On Monday 19 January 2009 09:34:43 am Zeeshan Zakaria wrote:
The reason why I introduced h priority here is that I needed to get the
variable CDR(duration) for DeadAGI script which I am also running in h
priority. Without h priority, I was getting correct CDR(dst) value but not
correct
On Mon, Jan 19, 2009 at 04:30:37PM +, Jared Smith wrote:
On Mon, 2009-01-19 at 11:10 +0100, Lukas Rypl wrote:
I am missing any description of zaptel/DAHDI alarms. The TE200 series
user manual contains only a description of LEDs states. These alarms
states are visible in
On Mon, 19 Jan 2009, Joseph wrote:
On Mon, 19 Jan 2009, bilal ghayyad wrote:
Hi All;
Anyone knows an IAX IP Phone works fine and tested?
Does polycom support IAX IP Phone?
Regards
Bilal
How about IAX2 adapter from digium?
I've been uing it and it works very well.
Wow, that has
I have an interesting observation which I thought I'd pass along to save
other people from spending time trying to 'fix' it.
One of my clients uses Charter's so called business phone service.
They provide 'analog' phone lines over IP. In general, they've worked
OK. End users were saying that
Thanks for this info. I am using Asterisk 1.4. I'll try this method and hope
it'll solve my problem in h priority.
On Mon, Jan 19, 2009 at 12:18 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Monday 19 January 2009 09:34:43 am Zeeshan Zakaria wrote:
The reason why I
Laurent a écrit :
Le 19.01.2009 08:50, Benoit a écrit :
Laurent a écrit :
Well, the telcos techs said a straight cable should do the trick, but
since i didn't get any isdn link up
with the straight, i built a crossover like what you described, with no
luck either.
My understanding is that Charter 'telephone' doesn't use IP at all but rather
uses some additional frequency spectrum on their cable network. Hence, the
reason why faxing with their service is reliable unlike other providers who are
*actually* using VoIP.
It sounds like they're suffering from
bilal ghayyad wrote:
Dear David;
At what price u get it?
Did u test it with IAX and SIP? Are u sure it is good? As really I did not
deal with chinese phone until now and I found it fine.
Regards
Bilal
--- On Mon, 1/19/09, David da...@linuxcrazy.com wrote:
From: David
Tim:
Are you referring to the older-style cable telephony where they had an
analog carrier on the cable plant, or PacketCable VoIP?
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent:
I have about 5 incoming USA SIP lines, but my provider does not have any sort
of roll-over or huntgroup feature. Does anybody have an idea on how I can
create a general number that will ring to the next available, non-busy SIP line
that I have? Is there a provider out there that would do
snip
My understanding is that Charter 'telephone' doesn't use IP at all but
rather uses some additional frequency spectrum on their cable network.
Hence, the reason why faxing with their service is reliable unlike other
providers who are *actually* using VoIP.
/snip
I think what you're referring
- David Gibbons d...@videon-central.com wrote:
I think what you're referring to is the general hesitance of the cable
providers to call their phone service VOIP service. VOIP still has a
negative connotation with most regular folks, so they don't want to
negative PR.
True.
I'm don't
On Mon, 19 Jan 2009, Jeff LaCoursiere wrote:
On Mon, 19 Jan 2009, Joseph wrote:
On Mon, 19 Jan 2009, bilal ghayyad wrote:
Anyone knows an IAX IP Phone works fine and tested?
How about IAX2 adapter from digium? I've been uing it and it works very
well.
Wow, that has NOT been my
Hello VoIP guys
Sorry for being somewhat off-topic. At the moment I am studying
informatics in the seventh semester and I need to start thinking about
my thesis. As I am very interested in VoIP technologies I thought about
picking this as my main topic. So far I have only little experience in
sp4rc wrote:
An idea: contact synchronisation via SIP
Are there any (working or concept) extensions on using SIP to synchronize
contacts
in the way icq does it? (server-side contacts)
The OpenSER/Kamailio/OpenSIPS technology stack along with XCAP and XMPP
provides pretty good solutions
Thanks!
I've just ordered a Linksys SPA8000 to try it out and compare it with my
Grandstream GXW4008 devices.
They are similar feature-wise. Linksys/Cisco should theoretically be a lot more
stable/reliable...
The only thing I'm missing in the SPA8000 but is available in the GXW4008
devices
I investigated Charter for our business phone systems and asked many of
these questions of the sales person. I was told they have a dedicated
part of the bandwidth available that is used just for phone traffic.
I could break out my college networking book and get you the frequency
break down
Digital Phone Service is a Fancy Marketing term Meaning Expensive VoIP
http://ezinearticles.com/?Digital-Phone-Service-is-a-Marketing-Term-for-Relabled,-Expensive-VoIPid=262018
Pure VoIP vs. Telephone and Cable VoIP
http://www.tmcnet.com/news/2006/08/16/1809766.htm
A telephone call over IP is
hi my friend,
i have to start a new company to provide asterisk Installation /
configuration to Small / mediom business
i'm looking for a asterisk expert to start with me
salary: 50%
i have a online store and is ready to use
please Call Me for mor informations:
Make a Sip Call
snip
I'd be willing to bet *TWO* pennies that you're correct. I certainly was not
coming into the conversation as an expert, just stating what I'd read/heard of
their service... hence the My understanding is that... beginning to the
email. :-)
/snip
Fair enough. I get worked up when I hear the
One problem to overcome is that your competitors are:
1) Literate.
2) Post to the right mailing lists.
Meftah Tayeb wrote:
hi my friend,
i have to start a new company to provide asterisk Installation /
configuration to Small / mediom business
i'm looking for a asterisk expert to start
snip
One problem to overcome is that your competitors are:
1) Literate.
2) Post to the right mailing lists.
Meftah Tayeb wrote:
/snip
Ha ha ha ha.
So, you're saying you don't want the job?
LOL.
___
-- Bandwidth and Colocation Provided by
David Gibbons wrote:
snip
One problem to overcome is that your competitors are:
1) Literate.
2) Post to the right mailing lists.
Meftah Tayeb wrote:
/snip
Ha ha ha ha.
So, you're saying you don't want the job?
LOL.
Well, actually, it would've been more proper and
Alex Balashov wrote:
One problem to overcome is that your competitors are:
1) Literate.
2) Post to the right mailing lists.
That'd be 2 me thinks.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty
I have been installing Asterisk as a SIP only system (no Digium Hardware)
for demonstration purposes. SIP users can connect to menus and voicemail
fine but the audio quality is terrible. The stock voicemail problems are bad
but basically understandable - voice menus recorded through the
Brian Alexander wrote:
I have been installing Asterisk as a SIP only system (no Digium
Hardware) for demonstration purposes. SIP users can connect to menus and
voicemail fine but the audio quality is terrible. The stock voicemail
problems are bad but basically understandable - voice menus
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Pierce
[pier...@westmancom.com]
Sent: Tuesday, January 13, 2009 5:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Mark,
Thanks - that was the problem I was having. Is there somewhere I could have
looked to have discovered the problem on my own? I would never have guessed
that on my own and my searches had not found it either.
Thanks again,
-Brian
On Mon, Jan 19, 2009 at 7:00 PM, Mark Michelson
On Mon, 19 Jan 2009, didier.cuffaut wrote:
2) From my first post, are these lines OK or wrong? (syntax error?)
tmsp = the delay in future.. say 100 seconds
exten= ra,n,System(NOW='date %S')
exten= ra,n,System(let NOW=$NOW+$tmsp)
exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec
Lukas Rypl wrote:
Hello,
I am missing any description of zaptel/DAHDI alarms. The TE200 series
user manual contains only a description of LEDs states. These alarms
states are visible in zttool/dahditool or in astersick CLI (zap show
status) and I wonder what is the real meaning of these
On Jan 19, 2009, at 12:35 PM, sp4rc wrote:
Hello VoIP guys
Sorry for being somewhat off-topic. At the moment I am studying
informatics in the seventh semester and I need to start thinking about
my thesis. As I am very interested in VoIP technologies I thought
about
picking this as my
2009/1/20 Olivier oza-4...@myamail.com
Hi,
Is anyone using Fring as a SIP client to an Asterisk server ?
Yes, testing it...
A prospective customer of mine is asking to integrate its iphones with an
Asterisk server and after googling, I still have some unanswered questions :
1. Which
2009/1/20 Zeeshan Zakaria zisha...@gmail.com
Hi everyone,
I googled this followed the instructions, but it hasn't work for me yet.
I have universal setting in SIPDefault.cnf and phone specific settings in
SIPXX.cnf. But it doesn't get registered.
I need to register it on two
On Jan 19, 2009, at 6:29 PM, D Tucny wrote:
2009/1/20 Olivier oza-4...@myamail.com
Hi,
Is anyone using Fring as a SIP client to an Asterisk server ?
Yes, testing it...
A prospective customer of mine is asking to integrate its iphones
with an Asterisk server and after googling, I
2009/1/20 John Todd jt...@digium.com
On Jan 19, 2009, at 6:29 PM, D Tucny wrote:
2009/1/20 Olivier oza-4...@myamail.com
Hi,
Is anyone using Fring as a SIP client to an Asterisk server ?
Yes, testing it...
A prospective customer of mine is asking to integrate its iphones
2009/1/19 Thomas Stein thomas.st...@knowledgetools.de
Hello.
Does someone know what order field means in followme.conf? The Doku says:
number= number to call[2nd #[3rd #]] [, timeout value in seconds [,
order in follow-me] ]
So an example would be:
number= 123124125,10,?
It would be
From my experience it won't register to the second box, only to the first
one. Why? god knows...
__Yehavi:
2009/1/20 D Tucny d...@tucny.com
2009/1/20 Zeeshan Zakaria zisha...@gmail.com
Hi everyone,
I googled this followed the instructions, but it hasn't work for
That's not my experience...
e.g.
SIP Phone show register
LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERED
line APR state timer expires proxy:port
--- - -- --
1 111 REGISTERED
On Tuesday 20 January 2009 06:17:14 D Tucny wrote:
number = 123124125,10,1 ; Would call 123, 124 125 first (all at the
same time as the same syntax in a Dial string would do), trying for 10
seconds
number = 126,10,3 ; Would call 126 third, trying for 10 seconds
number = 127,10,2 ; Would
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