Hi!
Did a quick test, worked as a clock:
exten = 0317998959,1,Set(CHANNEL(language)=se)
exten = 0317998959,n,Answer()
exten = 0317998959,n,ConfBridge(1001,s) 0317998959,n,Hangup()
On Thu, 18 Mar 2010 20:20:35 -0700, Kelvin Chan wrote:
Hi guys,
I'm trying to move away from meetme to
, Magnus Benngård wrote:
Hi!
We have alot of users who are having 2 phones, 1 fixed and 1
DECT.
I am looking for a way to log them into a queue and let both phone
rings. Let me try to explain:
0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a
queue.
queue add member SIP
/0317998975SIP/0317998985)
...then add Local/1...@agents to the queue.
On 03/14/10 00:03, Magnus Benngård
wrote:
Hi!
We have alot of users who are having 2 phones, 1 fixed and 1 DECT.
I am looking for a way to log them into a queue and let both phone
rings. Let me try to explain:
0317998975
Hi,
Did a test with Local, exten = 1234,1,Dial(Local/1...@agents)
[agents]
exten = 1,hint,SIP/0317998975SIP/0317998985
exten = 1,1,Dial(SIP/0317998975SIP/0317998985)
When calling 1234, both 0317998975 and 0317998985 rings
when answering in 0317998985, 0317998975 stops ringing, all fine but
Thx Rob!
On Mon, 15 Mar 2010 00:53:06 +1100, Rob Hillis wrote:
Glad to see I was able to point you in the right direction.
On 03/14/10 23:56, Magnus Benngård wrote:
queue add member Local/1...@agents to 0317998989 penalty 1 as Magnus
Benngard state_interface hint:1...@agents
On Sun
Hi!
We have alot of users who are having 2 phones, 1 fixed and 1 DECT.
I am looking for a way to log them into a queue and let both phone rings.
Let me try to explain:
0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue.
queue add member SIP/0317998975 to 0317998989
I am running Asterisk trunk with ooh323 towards an Avaya CM 3.1 without
any problems. I need your ooh323.conf and all relevant CM config
(signal-group, trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote: I'm trying to
connect an
hmmm... will be hard to help u without u having access... will do my best.
Here is my ooh323.conf anyway...
sip:/etc/asterisk# cat ooh323.conf
[general]
bindaddr=213.88.138.183
port=5088 --
_
-- Bandwidth and Colocation
and accept from both.
Step 2 could be round-robin send if both are up and alive...
Btw, did try trunk version, no support for multiple SRV records there.
Am 02.03.2010 08:50, schrieb Magnus Benngård:
Hi,
Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No
problem to get
Hi,
Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem
to get outgoing calls to work but i have some problems with incoming.
Did set srvlookup=yes in sip.conf. Sending all outgoing calls to
sip-corporate.tele2.se which is either sip-corporate1.tele2.se
(130.244.190.42)
Yes, when I added t38pt_usertpsource=yes to the NAT'ed fax everything
works!
Big thanks Johann!
On Sun, 21 Feb 2010 17:22:40 +0100, Magnus Benngård wrote:
t38pt_usertpsource=yes seems to do the trick, switches to T38 and fax
seems to go through (cant be 100% sure, the fax i am sending
Running Asterisk trunk with Siemens Gigaset S685IP, no normal problems,
just some with connected-line, probaly me, who is not smart enough. :(
Sound is great, use them both at our WAN and NAT'et at my home, DTMF
working as a clock... what more can I say?
On Mon, 22 Feb 2010 16:43:04 -,
Steinwendtner wrote:
Magnus Benngård wrote:
Gentlemen,
I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk.
0851711201 and 0851711290 is on our WAN, no NAT.
0197673581 is outside our WAN and needs to be NAT'ed.
Sending a fax from 0851711201 to 0851711290, no problem, switches
Gentlemen,
I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk.
0851711201 and 0851711290 is on our WAN, no NAT.
0197673581 is outside our WAN and needs to be NAT'ed.
Sending a fax from 0851711201 to 0851711290, no problem, switches to T38
and fax goes through.
Sending a from
Gentlemen,
Did tryout CONNECTEDLINE function, was exactly what I have been looking
for. But there are at least one thing I cant figure out.
Did a very simple and stupid extension 0317998955 and ran a test.
My phone (0317998975) dials 955, the display on my phone changes from
955 to
Gentlemen,
I did borrow an Aastra RFP 32 for some tests that i wanted to do.
Everything seems to be working except CLID. Setup as below:
DECT handset - GAP - Aastra RFP-32 - SIP - Asterisk - SIP Phone
When SIP Phone calls DECT handset, the display on the DECT handset only
shows the number of
checkout ${BLINDTRANSFER}
On Tue, 26 Jan 2010 15:48:51 +, Örn Arnarson wrote: Hi guys,
I am wondering (and have been unable to find out thus far) whether
Asterisk sets some special channel variables or something when a call is
transfered with the REFER method. Basically, I'm trying to
Morning,
Have some questions regarding receiving and sending faxes...
1:st example:
exten = 101,1,Answer()
exten = 101,2,Wait(3)
exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff)
exten = 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff
/var/spool/asterisk/tmp/fax.pdf)
exten
This is the setting i am using for Avaya CM to Aseterisk. (and pinf code
is working when dialing from Avaya to Asterisk
conference)sip:/etc/asterisk# cat ooh323.conf
[general]
bindaddr=213.88.138.183
port=5088
context=inputinterior.se
dtmfmode=rfc2833
;h323id=may day
;callerid=may day
Make sure u have the correct DTMF over IP (or what it is named in IP
Office, thats the CM name) setting on the signal-group. In my case: DTMF
over IP: rtp-payload
On Wed, 20 Jan 2010 16:11:58 -0800 (PST), hin lee wrote: Beside the
port number and the alaw, the only difference is the dtmf. I
Hi!
Any familiar with Avaya handsets? Did convert a 9650 handset to SIP. Cant
get the name just the number on the Avaya display.
Did put: SET DISPLAY_NAME_NUMBER 1 in 46xxsettings.txt
When I call from 0317998985 (Siemens DECT) to 0317998975 (Avaya 9650) i
just se 0317998985 in the Avaya
Is it in the trunk version or will it be added there?
On Tue, 22 Dec 2009 08:12:40 -0600, Kevin P. Fleming wrote:
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=Stefan
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
Hi!
Probably me that cannot read the manual...
I am trying to get all Keys that belongs to a certain Family
from the manager interface. Can just get single values for example:
Action: DBGet
Family: CFIM
Key: 0317998975
I was looking for something like Action: DBShow Family: CFIM.
Any one has
Hi!
I am trying to figure out how to rewrite calling number for all
extensions.
What I am trying to do is:
1) Have a block of rewriting rules that will apply to all calls:
Something like...
(???),ExecIf($[${CALLERID(num)}=702221448]?Set(CALLERID(all)=Cecilia
Benngard))
Did found a way to do it:
exten =
975,1,ExecIf($[${DB_EXISTS(CFIM/0317998975)}]?Goto(${DB(CFIM/0317998975)},1)
exten = 975,2,Goto(set-caller-id,s,1)
exten = 975,3,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
..
Hi!
Trying to understand how wrapuptime is working...
I have written a small php script that let agents log in/out
off a queue. That part is working as a clock but wrapuptime
is not doing what I expect.
Input Interiör - Queue Manager
0317998989 has 0 calls (max unlimited) in 'rrmemory'
)=0317998975))
exten = 977,n,ExecIf($[${CALLERID(num)} =
1234]?Set(CALLERID(num)=317998977))
exten = 977,n,ExecIf($[${CALLERID(num)} =
5678]?Set(CALLERID(num)=317998978))
[..]
exten = 977,n,Dial(SIP/0317998977)
On Mon, Dec 14, 2009 at 12:21 PM, Magnus Benngård
wrote:
Hi!
Trying to figure out how
Thx!
Did try callcounter=yes and it worked the way u told me!
It might have solved another problem 2, need to do some more tests...
On Sun, 13 Dec 2009 15:14:22 -0500, Leif Madsen wrote:
Philipp Kempgen wrote:
Magnus Benngård schrieb:
Set
call-limit=10
(or any other value 0)
Actually
.
On Sun, 13 Dec 2009 14:25:39 +0100, Magnus Benngård wrote: Hi!
Trying to figure out what I am doing wrong...
1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256)
1 Cell phone 00733025975 attached through H323.
extensions.conf
exten = 975,1,Goto(975-${DEVICE_STATE(SIP
Hi!
Trying to figure out how to rewrite calling number of an incoming call...
A cell phone (0733025975) dials a X-Lite (977).
X-Lite shows 733025975 at the display, but I want it to be 0317998975.
I thought i could do something like:
exten = 977/733025975,1,Set(CALLERID(number)=0317998975)
Thx, that did the trick!
On Sat, 12 Dec 2009 17:34:19 +0100, Philipp Kempgen wrote:
Magnus Benngård schrieb:
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
Set
call-limit=10
(or any other value 0)
extensions.conf
exten = 0317998975,hint
Hi!
Have a weired problem with Avaya 9650 phones:
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 0317998975,2,Hangup()
exten = 0317998975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
exten =
Hi!
Trying to figure out what I am doing wrong...
1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256)
1 Cell phone 00733025975 attached through H323.
extensions.conf
exten = 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten =
Hi all!
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
type=friend
regexten=0317998975
secret=
username=0317998975
callerid=Magnus Benngard
mailbox=0317998...@inputinterior.se
host=dynamic
canreinvite=yes
dtmfmode=rfc2833
nat=yes
disallow=all
Hi!
Avaya has just released SIP 2.5 which supports 9650 so i did convert one
from
H.323 to SIP and would like to share what I have to do to get basic stuff
working.
sip.conf
[0317998977]
type=friend
regexten=0317998977
secret=1234
username=0317998977
callerid=Stefan Andersson
Did a quick try, but I am said to say that I lack some setup info.
In manager.conf
enabled = yes
webenabled = yes
port = 5038
...
[abcti]
secret = secret
.
read =
system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
write =
Hi!
What version of spandsp is recommended to use when u compile
asterisk-trunk?
Best regards
MAGNUS BENNGRD ___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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To UNSUBSCRIBE or update options visit:
Hi!
Would be a very nice feature for example the following scenario:
Me has 2 phones, one ordinary SIP phone attached to the SIP server and one
Cell phone.
If someone calls my extension it will ring in both, but if I talk in for
example the SIP phone I dont want it to ring
on my cell phone.
I
Gentlemen,
Forgive me if I am posting at the wrong place!
I was going to test the new chan_ooh323 driver so I did install:
debian: Linux sip2 2.6.26-2-686 #1 SMP
dahdi-linux-complete-2.2.0.2+2.2.0
Asterisk SVN-trunk-r231692
Did enable chan_ooh323, everything compiled without any problems.
Hi!
Have probably not understand how fax is working in Asterisk 1.6.
I did install:
ptlib-v1_12_0
h323plus-v1_19_7
dahdi-linux-complete-2.2.0.2+2.2.0
spandsp-0.0.5
asterisk-1.6.2
asterisk-addons-1.6.2
make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax
But core show
to the Asterisk server.
Need to work on the T.38 and H.323 in the Asterisk, had to disable T.38 in
the Avaya CM to be able
to get the fax through. Can keep u posted if u are intrested in how it
goes.
On Mon, 30 Nov 2009 09:32:14 +0100, Magnus Benngård wrote: Hi!
Have probably not understand how fax
I am doing what u wanna atm but instead of an Alcatlet with SIP support i
have to
struggle with an Avaya CM without SIP but with H.323.
So far putting a trunk over Ethernet with SIP is the way I gonna go.
I havent run in to any show-stopper so far with my CM H.323 - Asterisk
integration.
On Mon,
On Sun, 22 Nov 2009 15:38:00 +0100, Leif Neland wrote: Magnus Benngård
skrev: Hi!
Part of extensions.conf:
exten = 985,1,Dial(SIP/0317998985H323/00702221...@avaya,20)
exten = 985,2,Goto(985-${DIALSTATUS},1)
exten = 985-BUSY,1,VoiceMail(0317998...@inputinterior.se,b [1])
exten = 985
Hi!
I did installed a Swedish voice prompts package, and added:
language=se to [general] section in sip.conf.
A SIP endpoint calling a conference get Swedish voice prompts but a call
that
comes through a H.323 trunk got English voice prompts. :(
I did try to add:
language=se to [general]
Hi!
That did the trick! Thx m8!
exten = 959,1,Set(CHANNEL(language)=se)
exten = 959,2,MeetMe(959)
exten = 959,3,Hangup()
On Fri, 20 Nov 2009 09:26:50 -0600, Tilghman Lesher wrote:
On Friday 20 November 2009 08:21:05 Magnus Benngård wrote:
Hi!
I did installed a Swedish voice prompts
Gentlemen,
I am trying to find a solution for running a VX-510 over SIP.
I know they have a BTB box that u can use for that purpose but it is, at
least in Sweden,
very expensive.
What I would like to do is something like below.
VX-510 -- SPA2102 -- Asterisk --H.323 trunk-- Avaya CM -- PSTN
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-
FROM: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] ON BEHALF OF Magnus
Benngård
SENT: Sunday, November 15, 2009 7:29 AM
TO: asterisk-users
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