Dear Asterisk community,
Yesterday, Russell Bryant finally made up his mind and confirmed on the
asterisk-dev mailing list that the next release of Asterisk will be 1.8, which
will also be a Long Term Support (LTS) release. This also means that the 1.4 is
now officially classed as a LTS
On Tue, Dec 22, 2009 at 11:41:35AM +0500, ABBAS SHAKEEL wrote:
Hello
When ever i try to use Dial DAHDI / SIP i get the following warning and
nothing happens and Asterisk moves to next instruction
Even i know that channel is free no one else is using it
[Dec 22 12:43:39] WARNING[11915]:
Hi C F,
I tried but does not work. It seems that my telco (telecom) does not
accept any number with a leading '*'.
Asterisk CLI returns busy:
empty_chan_in_stack: cannot empty channel 255
as if the channel were busy...but it works if I connect a normal phone
(and it worked with the old analog
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com wrote:
And what is the output of the ./configure? Does it generate any errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
Try tcpdump to see where RTP go from asterisk.
Configure your x-lite
Use stun server ?
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé :
It is a nat problem
François BERGANZ
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À :
Doug,
It doesn't respond to the INVITE - the trace says No response to the
INVITE?. If the phone doesn't even ring it's probably not getting anything,
which points to a problem with the router it's behind. How is the router set
up to deliver SIP and RTP to the phone?
On Tue, Dec 22, 2009 at 5:33
Where is your definition of codecs ??
On Tue, 2009-12-22 at 11:26 +0200, zehra yildiz wrote:
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone.
The softphone can call the other one but I can' t hear any voice. My
configuration files are below:
Hi C F,
I solved the problem!! It was under my nose...
If you are interested the solution is here:
http://www.misdn.org/index.php/FAQ_chan_mISDN
The right section is: key pad elements
Giorgio Incantalupo
C F wrote:
You would have to create a dialplan for it.
If your provider expects *67
Thanks Tzaffir,
Acutally i reinstalled DAHDI Asterisk and every thing seem to work fine
now.
i am using TDM800P with 8 FXO ports. the Number wasnt busy and asterisk
server can recieve calls through that channel but cant use that channel to
dial out.
As the problem is solved :) so what left to
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com
wrote:
And what is the output of the ./configure? Does it generate any
Hi all,
We need to start obtaining some billing files from BT via a dial-up ISDN
connection and I'm wondering if Asterisk is capable of doing this?
I need to make an ISDN dial-up CHAP connection and, once connected, grab some
files over FTP. Currently, our Asterisk box is connected to an
Hi,
I have a problem..I have 3 agents active in my queue..How can I get list
off all my active agents who was available when a call come?
in my extension.conf :
(1)exten = 6501,n,Queue(${EXTEN},ntT,,,1)
(2)exten = 6501,n,Queue(${EXTEN},ntT,,,25)
after the first one running, if i set like
Have a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS
On Tue, 22 Dec 2009 10:57:36 +, Will Payne w...@teambadger.co.uk
wrote:
Hi all,
We need to start obtaining some billing files from BT via a dial-up ISDN
connection and I'm wondering if Asterisk is capable of doing
I recommend you follow the detailed install guide in this book and install all
the required support programs etc.
http://downloads.oreilly.com/books/9780596510480.pdf
Thank you for contacting Kesher Communications Ltd.
IT Maintenance Clients can now receive a
Thanks a lot Alec, I´ll check
2009/12/22 Alec Davis siva...@paradise.net.nz
straight from our 1.6.1 dialplan, don't know about 1.2.14.
exten = s,n,Set(LIMIT_WARNING_FILE=beep)
exten = s,n,Set(LIMIT_TIMEOUT_FILE=call-terminated)
;terminate after 1 hour, start beep warnings at 10 minutes,
Hi,
I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip
extension definition, when I set language, it is not reported in the
extensions_custom.conf file (eg language=xx).
Am I missing something or is it not the right way to set language?
BTW, is this a valid place for AsteriskNow
Hi!
Is it possible, when placing a call that u see the name of the extension
in your diplay?
For example, 2 sip.conf entries:
[971]
callerid=Stefan
[975]
callerid=Magnus
975 calls 971 today 975 sees 971 in the display but would like to se:
Stefan or just Stefan or...
/Magnus
I have a 'CONGESTION' Status with R2 protocol.
While testing this scenario sip GW--àAsterisk Digium E1 R2
ProtocolàCisco E1 R2 protocolàsip Gw
Find below my error and configuration ,where are the errors in my
configuration ?
Magnus Benngård wrote:
Is it possible, when placing a call that u see the name of the extension
in your diplay?
For example, 2 sip.conf entries:
[971]
callerid=Stefan971
[975]
callerid=Magnus975
975 calls 971 today 975 sees 971 in the display but would like to se:
Stefan 975 or just
Kevin P. Fleming wrote:
This is called Connected Party information display, and it will be in
Asterisk 1.8.
Wasn't this scheduled for 1.6.2?
Doug
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Greetings,
Asterisk 1.6.2.0 was released last week. It's time to revisit release
plans for both current and future Asterisk releases.
For the past few months, there have been discussions regarding some
updates to Asterisk release policies. You can find my original -dev
list post on this
Hello,
Our asterisk is connected to an ericsson pbx by PRI.
What i want is the asterisk clients should call operator numbers by dialing 0
But, when a call is made to ericsson via number 0, it assumes that the
call is made from outside, so it doesnt allow to be dialed.
There are 3 real operator
Dan Journo wrote:
I recommend you follow the detailed install guide in this book and
install all the required support programs etc.
http://downloads.oreilly.com/books/9780596510480.pdf
*Thank you for
I am trying to track inbound and outbound calls by user. In sip.conf, I can
add an account code so that all outbound calls from user1 have that as the
accountcode in CDR, so that works fine. For inbound, if someone calls user1
direct, I can set the account code in the dial plan like this and it
On Tue, Dec 22, 2009 at 9:08 AM, Khaled W Chehab kche...@xplorium.comwrote:
I have a 'CONGESTION' Status with R2 protocol.
While testing this scenario sip GW--àAsterisk –Digium E1 R2 ProtocolàCisco
E1 R2 protocolàsip Gw
Find below my error and configuration ,where are the errors in my
the machine will lock up because the TDM board or the Dahdi
driver goes south. /var/log/messages starts filling up with repeated
messages:
kernel: TDM PCI Master abort
Thank you to everyone who has taken the time to reply.
First I am going to apply the fix what you know is broken
Huge thanks for mentioning what type of channel you are using.
On Tue, Dec 22, 2009 at 5:11 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi C F,
I solved the problem!! It was under my nose...
If you are interested the solution is here:
serach the option en sip.conf:
externip = you public ip
localnet=tus direcciones locales (address local)
saludos
Roman
On Tue, Dec 22, 2009 at 4:26 AM, zehra yildiz zyildi...@gmail.com wrote:
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can
At 5:01 PM on 22 Dec 2009, Oguzhan Kayhan wrote:
Hello,
Our asterisk is connected to an ericsson pbx by PRI.
What i want is the asterisk clients should call operator numbers by
dialing 0
But, when a call is made to ericsson via number 0, it assumes that the
call is made from outside, so
Doug Lytle wrote:
Kevin P. Fleming wrote:
This is called Connected Party information display, and it will be in
Asterisk 1.8.
Wasn't this scheduled for 1.6.2?
I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
as best I can tell from looking over the source code
On 12/22/09 6:00 PM, Kevin P. Fleming wrote:
Doug Lytle wrote:
Kevin P. Fleming wrote:
This is called Connected Party information display, and it will be in
Asterisk 1.8.
Wasn't this scheduled for 1.6.2?
I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
as best I
On 12/23/09 12:23, Russell Bryant wrote:
Wasn't this scheduled for 1.6.2?
I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
as best I can tell from looking over the source code :-)
Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze.
Hi,
How asterisk distinguish whether the re-invite is for codec change or for
a session refresh? I know that it checks the session version and decides
the same. But even if session version is different from the initial invite
and but it has the same codec, asterisk identifies that it is a
Hello,
You are dialing 00223344 with what you show: DAHDI/g1/00223344
That is not a real PSTN number in any country as for as I know. Do you have
the proper outbound route setup? Is your outbound route stripping digits?!
-Bruce
On Tue, Dec 22, 2009 at 9:08 AM, Khaled W Chehab
On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote:
On 12/23/09 12:23, Russell Bryant wrote:
Wasn't this scheduled for 1.6.2?
I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
as best I can tell from looking over the source code :-)
Nope, it's not in 1.6.2.
why the edition jump from 1.6.2 to 1.8 , what's the reason? the number
of the edition always
confuse me.
2009/12/23 Tilghman Lesher tles...@digium.com:
On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote:
On 12/23/09 12:23, Russell Bryant wrote:
Wasn't this scheduled for 1.6.2?
I don't
On Wednesday 23 December 2009 01:31:04 Zhang Shukun wrote:
2009/12/23 Tilghman Lesher tles...@digium.com:
On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote:
On 12/23/09 12:23, Russell Bryant wrote:
Wasn't this scheduled for 1.6.2?
I don't believe so, but I could be mistaken. It's
Can I define the realm on a per peer basis ??
Can I define a realm to be used for one peer and another realm for
another peer in sip.conf ??
I have an ITSP that I need to authenticate with a realm that they set.
But this realm is not valuable for other peers.
Jonas.
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