Solved it with returning from Dial to Macro
exten = _X.,1,NoOp(SIPID1 = ${SIPCALLID})
exten = _X.,n,Dial(SIP/${EXTEN}@some_IP,120,M(to))
exten = h,1,NoOp
[macro-to]
exten = s,1,NoOp(${DIALSTATUS},1)
exten = s,n,Set(SIPID2=${SIPCALLID})
exten = s,n,Set(CDR(sipid2)=${SIPID2})
exten =
Do I need to modify chan_phone application to make it works or it is
available in net.
Thanks
Nikhil
On 02/17/2011 12:52 PM, Khaled W. Chehab wrote:
Install asterisknow and begin from there.
http://www.asterisk.org/asterisknow/
and don’t miss to read the documentation
Hi,
Thanks Pezhman, I have yet to start the configuration. I was just diong the
pre survey that if I get into some problem there are people who can help. I
will contact for any help required
/ag
On Wed, Feb 16, 2011 at 9:42 PM, Pezhman Lali l...@lopl.net wrote:
dear
I have a good exp in
Anyone had any experience of using this phone with asterisk ? Trying
to find out if I can provision it using tftp / http
Thanks
Julian
--
Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker
--
_
-- Bandwidth
On 17 Feb 2011, at 10:04, Nikhil wrote:
Do I need to modify chan_phone application to make it works or it is
available in net.
Why not use a proper sip client?
S
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-- Bandwidth and Colocation Provided by
Hi,
I'm excited to announce that the guys from AG Projects are stopping by
for a beer tomorrow on VoIP Users Conference, aka VUC.
You should already be familiar with their excellent multi-platform SIP
client, Blink (http://icanblink.com)
While Adrian and Saúl enjoy a few exotic brews with us,
Hi guys,
i am looking for application to modulate voice of speaker. This is
supposed to be FUN type of service, where user can call a premium number
and from IVR menu choose woman's or men's voice type and then call
friend to make him a joke :)
Is there such application in standard asterisk's
On Thu, Feb 17, 2011 at 12:02 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
No parameters were rejected. Maybe my perception of backlight off is
incorrect. When it is off I expect it so be similar to a Cisco 7961. So no
light whatsoever and very hard to read in dim light. Yet in the
Yeah it's the same thing; I think.
I think we have different config files.are you using the split?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Thursday, February 17, 2011 12:10 AM
To: Asterisk Users Mailing
I have ind.pattern.
Where in ind.pattern were you looking for to turn off MWI?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Thursday, February 17, 2011 12:02 AM
To: Asterisk Users Mailing List -
Hi to all.
I make some tests with Asterisk 1.8.2 in Realtime. But i have one problem,
the asterisk don't connect in the base and show this message:
[Feb 17 11:18:01] WARNING[19061]: res_config_mysql.c:441
realtime_multi_mysql: MySQL RealTime: Invalid database specified:
'asterisk_teste' (check
On Thu, 2011-02-17 at 11:28 -0200, Rodrigo Lang wrote:
Hi to all.
I make some tests with Asterisk 1.8.2 in Realtime. But i have one
problem, the asterisk don't connect in the base and show this message:
[Feb 17 11:18:01] WARNING[19061]: res_config_mysql.c:441
realtime_multi_mysql: MySQL
Hi,
I have an Asterisk 1.8.2.3 installed (public IP) with a peer (Polycom
IP601) installed behind NAT.
When the peer makes a call, it's working without any problem. But when a
call is coming back, it ends up with a Got SIP response 400 Bad
Request back from xx.xx.xx.xx where the xx.xx.xx.xx
In your extension, do you have nat=yes
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Tardif
Sent: Thursday, February 17, 2011 9:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Got SIP response 400 Bad
Hi everyone,
I am reading through Sangoma Wiki right now. But someone may already and
quickly notice this. I have a system that is down since the morning (maybe
power intruptions). All seems fine except for wanrouter status shows
disconnected. Following are the alarms raised. Should I call telco
On 02/17/2011 02:28 PM, ERIC HERRON wrote:
I have ind.pattern.
In firmware version 3.3.1? I have only found a reference to ind.pattern
in the Simplify Configuration Improvements Guide which you mentioned
yesterday. Afaict there is no ind.pattern section in the 3.3.0 Admin
Guide and the
On 02/17/2011 02:26 PM, ERIC HERRON wrote:
Yeah it’s the same thing; I think.
I think we have different config files…are you using the split?
Unfortunately I have no idea what the split means. Can you please explain?
Regards,
Patrick
--
On 02/17/2011 02:23 PM, Ryan Wagoner wrote:
[snip]
The color screen must be different or it is a firmware bug. Was it any
different on 3.2.x vs 3.3.x? On my IP550 you can still read the screen
I have not tried firmware 3.2.x. I'll give that a try once I figure out
the old config system.
Hi everyone,
I have fresh installation of e1 line with 15 B channels, but
unfortunately calls are not going thru.
Do you know what could be wrong with config at my end? Or maybe this is
something with telco? Any hint will do :)
I am getting following debug:
2011/2/17 Ishfaq Malik i...@pack-net.co.uk
On Thu, 2011-02-17 at 11:28 -0200, Rodrigo Lang wrote:
Hi to all.
I make some tests with Asterisk 1.8.2 in Realtime. But i have one
problem, the asterisk don't connect in the base and show this message:
[Feb 17 11:18:01] WARNING[19061]:
Dear, I always had one E1 card with one span, so I've never had any
problem in set it up through /etc/dahdi/sustem.conf and
/etc/asterisk/chan_dahdi.conf because I put span=1.
But now I have a PBX with two E1 cards with 4 span (8 span in total).
How do I have to define both card in system.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Sent: Thursday, February 17, 2011 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] calls are not going thru e1
On 17.02.2011 17:47, Danny Nicholas wrote:
snip
Post your dahdi show channels output.
Have you checked the lines with a regular handset?
here it is:
*CLI dahdi show status
Description Alarms IRQbpviol CRC4
Fra Codi Options LBO
T2XXP (PCI) Card 0 Span 1
system.conf:
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3
Hi all,
I have a few doubts regarding asterisk...
1. Is asterisk can be used as stateless and stateful proxy? If yes i have 4
questions...
1. How we can configure stateful proxy?
2. Where we can configure stateful proxy?
3. How does the stateful proxy stores the requests and responses?
Hi everyone,
I want to know if there is any way to pickup a call from an specific exten.
I know we can configure our pickup code on features.conf, and it works with
the members of our pickupgroup, but, if there are two different calls, it's
impossible to control which call we pickup.
Is there
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
viswavardhanreddy karna
Sent: Thursday, February 17, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Regarding Asterisk
Hi
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jordi Bou
Sent: Thursday, February 17, 2011 11:12 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Pickup from an specific exten
Hi everyone,
I want to
On 02/17/2011 06:06 PM, viswavardhanreddy karna wrote:
Hi all,
I have a few doubts regarding asterisk...
1. Is asterisk can be used as stateless and stateful proxy? If yes i
Afaik Asterisk is a B2BUA (google B2BUA if you don't know what it is)
and does not have proxy capabilities
On Thu, 2011-02-17 at 11:13 -0600, Danny Nicholas wrote:
__
From:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jordi
Bou
Sent:
On Thu, 2011-02-17 at 11:12 -0600, Danny Nicholas wrote:
...Please help me as of i am doing my master thesis on evaluating the
performance of various open source projects
Thanks in advance
Awaiting for the reply as soon as possible,
Awesome. Any institution that issues a
On 11-02-17 12:06 PM, viswavardhanreddy karna wrote:
Hi all,
I have a few doubts regarding asterisk...
You are correct, Asterisk is a B2BUA, not a SIP Proxy.
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at:
On 11-02-17 07:47 AM, Albert wrote:
Hi guys,
i am looking for application to modulate voice of speaker. This is
supposed to be FUN type of service, where user can call a premium number
and from IVR menu choose woman's or men's voice type and then call
friend to make him a joke :)
Is
On 17.02.2011 20:21, Paul Belanger wrote:
On 11-02-17 07:47 AM, Albert wrote:
Hi guys,
i am looking for application to modulate voice of speaker. This is
supposed to be FUN type of service, where user can call a premium number
and from IVR menu choose woman's or men's voice type and then
*A(**x**)*: Play an announcement (*x*.gsm) to the called party.
2011/2/16 Faisal Hanif fai...@vopium.com
You can do it using callback files or AMI.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Songtao Yu
*Sent:*
On Sat, Feb 12, 2011 at 07:31, ast guy ast...@gmail.com wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone that works well with
asterisk server.
Polycom phones are still working well and durable as a brick.
On 02/17/2011 12:27 PM, Mike wrote:
Hi,
Is there ANY way for me to see the status of the Polycom DND buttons in
the Asterisk hints? I`m using the BLF buttons to see the status of other
people`s lines, and DND should logically be somehow reflected (I don`t
care as much about Polycom showing the
On 18.02.2011 00:30, Andrew Joakimsen wrote:
On Sat, Feb 12, 2011 at 07:31, ast guy ast...@gmail.com wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone that works well with
asterisk server.
Polycom phones are
On 02/17/2011 05:51 PM, Albert wrote:
On 18.02.2011 00:30, Andrew Joakimsen wrote:
On Sat, Feb 12, 2011 at 07:31, ast guyast...@gmail.com wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone that works well with
On 18/02/11 1:00 PM, Kevin P. Fleming wrote:
On 02/17/2011 05:51 PM, Albert wrote:
Linksys SPA921, SPA922, SPA941, SPA942 are also working pretty well.
... and have all been discontinued by Cisco.
Kinda, they've pretty much just rebadged them Cisco SPA303 etc - all the
same options in the
If you already have experience with linux asterisk will be easy for you.
Other people will reply with official links but here is how I use Asterisk in
my small home office www.cognation.net/asterisk
Cheers,
Dean
From:
Hello everybody,
Can someone explain [gGrR] in Dial() function?
To dial external extension 18005551212 over channel 2 we will use:
Dial(DAHDI/2/18005551212)
To dial external extension 18005551212 over one of channel from group of
channels (nr 2) we will use:
Dial(DAHDI/g2/18005551212)
So lets
On 11-02-17 07:31 PM, Albert wrote:
Hello everybody,
Can someone explain [gGrR] in Dial() function?
*CLI core show application Dial
To dial external extension 18005551212 over channel 2 we will use:
Dial(DAHDI/2/18005551212)
To dial external extension 18005551212 over one of channel
Aloha,
We have added the ability to dynamically forward or send a voicemail
to more than one mailbox.
Here is the link -
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18835
There is a diff file and a drop-in replacement for app_voicemail.c.
Here are some basic instructions for
On Thursday 17 February 2011 14:13:04 Albert wrote:
On 17.02.2011 20:21, Paul Belanger wrote:
On 11-02-17 07:47 AM, Albert wrote:
Hi guys,
i am looking for application to modulate voice of speaker. This is
supposed to be FUN type of service, where user can call a premium
number and
On Fri, 2011-02-18 at 00:51 +0100, Albert wrote:
On 18.02.2011 00:30, Andrew Joakimsen wrote:
On Sat, Feb 12, 2011 at 07:31, ast guy ast...@gmail.com wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone
Hi List,
Were upgrading our network switches and need to create multiple VLAN groups,
but since our Squid Proxy (Transparent Proxy) Server should be accessible to
all VLAN groups we need to setup a trunk grouping inside our Squid Proxy
Box. Is anyone has a documentation or code on how to
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