Hi Ira,
for Aastra phones I have done this application to resolve busy/xfer
transfer:
extensions.conf
===
exten = _1X,1,GotoIf($[${SIPPEER(${EXTEN}|curcalls)}1]?free:busy)
exten = _1X,n(free),Dial(SIP/${EXTEN},,tTr)
exten = _1X,n,Hangup()
exten
I am working in a university also , and nowadays, we are aking some tests
to start using asterisk in some areas of our campus. Because it costs a
lot more cheper than extending our PBX system.
It seems ok for us to make a hybrid system in the campus area which should
be about 1000 clients for the
hi, all
asterisk 1.4.24 , zaptel 1.4.10.1 , E1
Manager API Action :
Action: Originate
Channel: ZAP/G1/888
Callerid: 12345678
Context: callout
Exten: s
Priority: 1
extensions.conf
[callout]
exten = s,1,Answer()
exten = s,n,Wait(10)
exten = s,n,Hangup()
when the phone
There are various ways of doing this.
You could use an analogue port/ATA and connect any good old fashioned
intercom to it (Pantel are a good make).
You can now get SIP intercom systems as well. I haven't tried on of
these - but they look good (and can contain a camera as well if needed).
HTH
John Knight a écrit :
make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 »
WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers
is missing; modules will have no dependencies and modversions.
specifically Symbol version dump
Tzafrir Cohen a écrit :
On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote:
Hi,
We installed the latest 1.4.24 on a test machine and can't get zaptel
nor dahdi compile. It's a Linux Debian Etch. Errors we have:
keewi:/usr/src/dahdi-linux-2.1.0.4# make
make -C
Hi,
If anyone has the same problem, I solved it doing:
genzaptelconf -sdv
It might have been a problema with the card or the module.
Regards
Imanol Pardavila escribió:
Hi,
I stilll continue with the problem but I have noticed something new
that maybe a clue. The noise during the call
Hi,
Would you please let me know the performance of asterisk realtime in case I
will have millions of SIP users?
Thanks,
Krunal Patel
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Le'an Liu wrote:
My questions:
1. G.726 16/24/32/40 supported in asterisk-1.6.0.5?
No. Only G726-32 is supported in all Asterisk versions.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber:
Stefan Schmidt wrote:
hello, you could retrieve the config from you SPA with the following
url: http://ipofyourphone/admin/spacfg.xml .
That works well with the Linksys phones, but not with the SPA-3102 which
isn't really a phone, but an ATA. My 3102 has software version 5.1.6.
/Per
Hi,
I'm having problems when the callerid of a user defined in the
sip.conf contains special characters such as: ñ, á, é, í, ó , etc. The
strange thing is that these characters are displayed correctly in the
dialplan by using the sip show peer command, but if this user makes a
call, these
Krunal Patel wrote:
Hi,
Would you please let me know the performance of asterisk realtime in
case I will have millions of SIP users?
I don't think it will work on a single server, with or without realtime.
If only a very small amount of them would be online at any moment, maybe
it will work
How does a Push-to-talk intercom interface with Asterisk?
Andrew Thomas wrote:
There are various ways of doing this.
You could use an analogue port/ATA and connect any good old fashioned
intercom to it (Pantel are a good make).
You can now get SIP intercom systems as well. I haven't
IMHO when users scale up to such levels, Asterisk falls short, I made a c
ouple large implementations and the best approach is using OpenSer as SIP
engine (along with his own media proxy if required by your network schema)
and use Asterisk as Vertical Services Provider, such as email, IVR, in
Have a look at http://www.northsupply.co.uk/ (under Door Access
Systems).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris
Mason (Lists)
Sent: 18 March 2009 11:56
To: Asterisk
2009/3/18 Kevin P. Fleming kpflem...@digium.com
Le'an Liu wrote:
My questions:
1. G.726 16/24/32/40 supported in asterisk-1.6.0.5?
No. Only G726-32 is supported in all Asterisk versions.
Perhaps the confusion in the voip-info page mentioned is due to the other
G726 rates being supported
On Wed, 18 Mar 2009, Chris Mason (Lists) wrote:
How does a Push-to-talk intercom interface with Asterisk?
I think the generic answer is expensively.
If Xorcom made just the IO part of their channel banks then it might be
cheaper, however ...
What I've seen so-far is an intelligent box with
On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno tipas...@gmail.com wrote:
I have a weird problem with call using my T1 card. I can make calls fine
using my analog and IP phones, but when I try to initiate a call using a
.call file, I get the following error
-- Attempting call on
Hi,
I have 2 instances of asterisk running and taking to common mysql
database via ODBC connection.
I am facing some issue while running bulk call (around 100 calls at a
time),like few call gets error out of 100, i am suspecting that SQL query is
failing and the error i get in one the call
Nope, I always dial 1 + 10 digits for all my numbers. It works on all
numbers when I am using my phone (Analogue or IP) but when I do it using a
.call file it does not work on some numbers mostly. That is the weirdest
thing I have ever seen. I tried different codecs in the call file, I still
Hi,
I have 2 instances of asterisk running and taking to common mysql
database via ODBC connection.
I am facing some issue while running bulk call (around 100 calls at a
time),like few call gets error out of 100, i am suspecting that SQL query is
failing and the error i get in one the call
Is there a way to set/clear a BLF LED on a phone from the dialplan?
I want to use one as an indicator of some state in the PBX - in this case
it's night mode but I can think of other applications.
I have BLFs working just fine for normal stuff, just wonderin if I can
use them for more!
Gordon Henderson wrote:
Is there a way to set/clear a BLF LED on a phone from the dialplan?
I want to use one as an indicator of some state in the PBX - in this case
it's night mode but I can think of other applications.
I have BLFs working just fine for normal stuff, just wonderin if I
Hello,
I am running asterisk 1.4. For argument's sake I have 4 telephones. 2
support video, 2 do not.
Calls between phones work fine and codecs are properly negociated. I
have videosupport=yes in sip.conf and when the two video phones
communicate I have video.
I call meet me with this
Hello,
I am running asterisk 1.4. For argument's sake I have 4 telephones. 2
support video, 2 do not.
Calls between phones work fine and codecs are properly negociated. I
have videosupport=yes in sip.conf and when the two video phones
communicate I have video.
When the video phone calls the
On Wed, 18 Mar 2009, Dave Fullerton wrote:
Gordon Henderson wrote:
Is there a way to set/clear a BLF LED on a phone from the dialplan?
I want to use one as an indicator of some state in the PBX - in this case
it's night mode but I can think of other applications.
I have BLFs working just
Hi all,
Is there something like a global h exten, that gets called on every hang
up, no matter what exten?
Thanks,
Gabriel
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On Wed, 18 Mar 2009, Gabriel Ortiz Lour wrote:
Is there something like a global h exten, that gets called on every
hang up, no matter what exten?
(no matter what context)
Nope -- but it sounds like a great idea.
I do it this way...
I define an h template:
[h](!)
On Wed, Mar 18, 2009 at 11:57 AM, Steve Edwards
asterisk@sedwards.comwrote:
On Wed, 18 Mar 2009, Gabriel Ortiz Lour wrote:
Is there something like a global h exten, that gets called on every
hang up, no matter what exten?
(no matter what context)
Nope -- but it sounds like a great
On Mon, Mar 16, 2009 at 20:26, Marc Charbonneau timebandit...@gmail.com wrote:
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE)
I like the Polycom IP-330. 2 lines, nice
I concur on TelephonyDepot.com.
I really like the Grandstream GXP2000 @ about $95, and the Budgetone 200 at
about $48 bucks (No POE on the 200) but dual Ethernet ports.
Personally I stopped using the SNOM360 and use the GXP2000 with a headset.
Both of those Grandstreams support 2.5 mm headset
I've worked with VoIP Supply several times in the past. I've been very
pleased with their service. And if you compare the prices of the two phones
you mention: Polycom IP 320 330 a difference of 109.94 vs 106 and 84.95
vs 83 seems to disperse the allegation of being overpriced.
Thanks,
David
Tzafrir Cohen a écrit :
On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote:
Hi,
We installed the latest 1.4.24 on a test machine and can't get zaptel
nor dahdi compile. It's a Linux Debian Etch. Errors we have:
keewi:/usr/src/dahdi-linux-2.1.0.4# make
make -C
Hi,
I'm experiencing a quite strange behavior while trying to receive faxes
through Asterisk (either directly through app_rxfax or with spandsp +
hylafax).
Config:
HFC quad BRI card (3 T0 connected to the card)
Asterisk 1.4.21
asterisk-app-fax 0.0.20070624-2
hylafax 2:4.4.4-10.1
libpri 1.4.2
How do you require a password for a voicemail box? I have been
searching all day, and can't find any type of security setting for
voicemail. I am looking for some what to have some minimum security
like no blanks, can't be the same as the extension, can't be
sequential numbers or repeated
Greetings chan_mobile users,
I have just merged my refactor of chan_mobile into asterisk-addons trunk
and now the code needs testing. The changes I have made should improve
the stability and reliability of the code and should also improve audio
quality. Error reporting should be improved as
On 17/03/2009 9:10 a.m., Doug wrote:
This looks great! A few questions...
in the standard extension macro we add a line:
Is this in extensions.conf?
Yeah, we have a macro (which is in the default extensions.conf) which we
add that line to.
exten =
It's so uncommon for me fxs and fxo cards and based on the reference
of sip.conf files and accounts i totally missed last paragraph where
it was mentioned only analogue lines and fxs (phone).
my appologies.
E1 and BRIs and sip trunks have been overloading my last month of work.
cheers,
--
Marco
On Wednesday 18 March 2009 17:02:33 Jonathan Thurman wrote:
How do you require a password for a voicemail box? I have been
searching all day, and can't find any type of security setting for
voicemail. I am looking for some what to have some minimum security
like no blanks, can't be the same
On 18/03/2009 9:58 p.m., MaxGao wrote:
hi, all
asterisk 1.4.24 , zaptel 1.4.10.1 , E1
Manager API Action :
Action: Originate
Channel: ZAP/G1/888
Callerid: 12345678
Context: callout
Exten: s
Priority: 1
extensions.conf
[callout]
exten = s,1,Answer()
exten = s,n,Wait(10)
exten
On 19/03/2009, Jonathan Thurman jthurma...@gmail.com wrote:
Also, is there a way to retain deleted messages for a length of time
before they are purged? We currently have that feature on our
production VM server that I am trying to replicate. Thanks!
Could this be done with a simple
Any idea what this means? And why they are different?
-
Extension Changed 22142[default] new state Idle for Notify User 31001
(queued)
Extension Changed 22142[default] new state Idle for Notify User 30060
-
I have googled and searched, and can't find anything on this subject.
Does anyone
This has to be a bug, because I dont know what else to try here
On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno tipas...@gmail.com wrote:
Nope, I always dial 1 + 10 digits for all my numbers. It works on all
numbers when I am using my phone (Analogue or IP) but when I do it using a
.call
在2009-03-19?06:53:56，Matt?Riddell?li...@venturevoip.com?写道：
On?18/03/2009?9:58?p.m.,?MaxGao?wrote:
?hi,?all
??asterisk?1.4.24?,?zaptel?1.4.10.1?,?E1
??Manager?API?Action?:
?Action:?Originate
?Channel:?ZAP/G1/888
?Callerid:?12345678
?Context:?callout
?Exten:?s
?Priority:?1
On 19/03/2009 2:17 p.m., MaxGao wrote:
??2009-03-19?06:53:56??Matt?Riddell?li...@venturevoip.com???
On?18/03/2009?9:58?p.m.,?MaxGao?wrote:
?hi,?all
??asterisk?1.4.24?,?zaptel?1.4.10.1?,?E1
??Manager?API?Action?:
?Action:?Originate
?Channel:?ZAP/G1/888
oh, i am sorry, plain text :
hi, all
asterisk 1.4.24 , zaptel 1.4.10.1 , E1
Manager API Action :
Action: Originate
Channel: ZAP/G1/888
Callerid: 12345678
Context: callout
Exten: s
Priority: 1
extensions.conf
[callout]
exten = s,1,Answer()
exten = s,n,Wait(10)
On Mon, Mar 2, 2009 at 3:07 PM, Brandon B. bran...@brellsystems.com wrote:
On Fri, Feb 27, 2009 at 7:49 PM, Jared Smith jsm...@digium.com wrote:
As I understand it, the LBO is effectively an attenuation value, with a
higher number meaning less attenuation. This way, you don't get too hot
On Wed, Mar 18, 2009 at 4:18 PM, Andrew Furey andrew.fu...@gmail.com wrote:
On 19/03/2009, Jonathan Thurman jthurma...@gmail.com wrote:
Also, is there a way to retain deleted messages for a length of time
before they are purged? We currently have that feature on our
production VM server
The AstLinux Team is happy to announce that AstLinux 0.6.4 is available.
All users of AstLinux are encouraged to upgrade since this release
fixes the recently reported security vulnerability in Asterisk 1.4.23.1
Right now a mix up on the Sourceforge site is preventing us from
uploading full
Hi All,
I have a working asterisk 1.4.23.1 on server.
OS: Centos 5.2
Suddenly asterisk has stopped to process calls crashed.
I found that asterisk has generated coredumps.
I have restarted asterisk it started to work as expected without any
issue.
Would you please help me out to troubleshoot the
2009/3/17 Marc Charbonneau timebandit...@gmail.com
I was looking at the aastra 9133i, however I was informed that this phone
is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE)
I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet,
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