Instead of picking from multiple scripts, send the action to the script
in a variable like the dial status
On 4/12/11 4:58 PM, Warren Selby wrote:
Sorry for the top post, on my phone...
It makes sense for someone who has written a custom visual voicemail
application to be able know when the
Maybe try removing the /301 from your register line and either use
nothing or try /sip.broadvoice.com
vijay.go...@alliance-infotech.com wrote:
Hi,
I am using Asterisk 1.4.25. I have one Broadvoice account. I Integrated
this broadvoice account with Asterisk Server.
I am Able to Make calls
I'm guessing that asterisk is connecting to the spa2012 through the
local ip. I would make sure you have canreinvite=no.
James Hankins wrote:
Greetings,
I'm having a heck of a time with one way audio on a SPA2012. It's
public IP connected directly to cable modem. One line configured.
It is already a macro, not sure about passing an array of numbers.
Alex Samad wrote:
On Wed, May 20, 2009 at 03:16:34PM -0400, M Hulber wrote:
Alex Samad wrote:
On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
[snip]
I left the busy after dial
Couldn't he also just do a sip set debug to view the responses coming
back?
Jeff LaCoursiere wrote:
On Wed, 20 May 2009, John Regal wrote:
Thanks for the reply and apologize for the double post. My original post
landed in another thread and thought it may have been missed...
I
Alex Samad wrote:
On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
What you have here should work just fine except:
exten = _1866NXX,1,Dial(ZAP/g1/${EXTEN}) -- note the change from n to 1.
I also don't understand why you have an Answer after your Dial statements.
I would
:)
Dimitris ...
M Hulber wrote:
I'm looking at the 942 and if you look under the EXT 'n' settings in the
'SIP Settings' section you can select SIP Transport. There are 3
options: UDP / TCP / TLS. I'm using the 6.1.5 software.
Dimitris Counalakis wrote:
Hi all,
I'm new
I'm looking at the 942 and if you look under the EXT 'n' settings in the
'SIP Settings' section you can select SIP Transport. There are 3
options: UDP / TCP / TLS. I'm using the 6.1.5 software.
Dimitris Counalakis wrote:
Hi all,
I'm new to this list, so forgive me if I'm not supposed to
What you have here should work just fine except:
exten = _1866NXX,1,Dial(ZAP/g1/${EXTEN}) -- note the change from n to 1.
I also don't understand why you have an Answer after your Dial statements.
I would do this:
; Outbound via POTS
[general-outbound]
include = pri_outbound
exten =
Are these attempts to scam SIP calls through my Asterisk server:
[May 13 22:50:41] NOTICE[30888]: chan_sip.c:17295 handle_request_invite:
Call from '' to extension '084312297134' rejected because extension not
found.
[May 14 13:36:35] NOTICE[30888]: chan_sip.c:17295 handle_request_invite:
Call
memory) of
each connection. Is it true? How about using madplay?
On 4/28/09, M Hulber asterisk-ad...@hulber.com wrote:
Didn't do mms but have implemented using Shoutcast. I have instructions
at the link below:
http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16
Didn't do mms but have implemented using Shoutcast. I have instructions
at the link below:
http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16/
Rilawich Ango wrote:
Hi,
I follow the
web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
-
Without having tried it I notice the output is x86-64 and not x86_64.
Could there be a typo somewhere?
sean darcy wrote:
1.6.1 svn 190575:
CC=cc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent menuselect
make[1]: Entering directory
I checked out the 190660 trunk and went all the way through make without
a problem.
Linux asterisk.hulber.com 2.6.18-128.1.6.el5 #1 SMP Tue Mar 24 12:05:57
EDT 2009 x86_64 x86_64 x86_64 GNU/Linux
--
Output through generating input for menuselect:
[r...@asterisk trunk]# ./configure
I've seen that message when then endpoint is not available.
Cary Fitch wrote:
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer
I followed the Ronald Lewis instructions and was able to get EC2 to run
Asterisk. I was able to use IAX2 so I'm not sure what you are saying.
You should also be able to build dahdi but of course you won't have any
physical devices in the machine. I think for meet-me dahdi provides a
Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault.
Reverted to 1.6.0.6 and back to normal.
--
Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24
EST 2009 x86_64 x86_64 x86_64 GNU/Linux
Apr 3 11:49:56 asterisk kernel: asterisk[3780]:
I believe it's echo and/or jitter being measured when the call is
connected as I recall it being explained. This issue has existed for a
long time and I'm not sure there's much you can do about it except to
wait for a second before speaking when a call is connected. I think
maybe I have
You've had some good suggestions so far but honestly the brute force
method is not that difficult. I have been in the process of trying to
make my dialplan more concise (fewer statements) but haven't tried to do
anything about this one:
; Toll-Free
exten =
Cary,
You also forgot 880, 881, 882 although I'm not sure I've ever even come
across one of those.
Cary Fitch wrote:
In my previous reply, I may be wrong, 877 is probably a valid toll free
NPA, add it in the mix.
Cary Fitch
-Original Message-
From:
Did these announcements stop coming on the [asterisk-announce] group? I
only seem to get sporadic announcements there.
Asterisk Development Team wrote:
The Asterisk Development Team is pleased to announce the first release
candidate of Asterisk 1.6.0.7, tagged as version 1.6.0.7-rc1. Release
Umm, I don't think a called number sends any callerid info as there's
probably not even a protocol for that. Maybe you need to post a sample
CDR. The only thing I could think of is if you are calling an internal
extension and asterisk is posting the callerid you have defined for that
on their phone instead of
going through the IVR hell.
Vikas
On Wed, Feb 25, 2009 at 3:13 PM, M Hulber asterisk-ad...@hulber.com wrote:
Since it's not clear from this thread of conversation, do you need 100
unique DIDs? If you do:
That NPA is owned by Pacbell with the central office
what legal statues setting caller-id fraudulently falls under?
Is there a federal law you can reference?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of M Hulber
Sent: Wednesday, February 25, 2009 4:13 PM
I agree with the comments on the intended target market for this phone.
In defense of Polycom, if your TFTP server is external you could connect
to a remote access point by setting up WEP/WPA fairly easily from
Starbucks or wherever you are. If it requires web authentication to get
through
Bob,
Ok, that's the route I ended up taking where all lines are the same
user. I put the AUTH an LINEn_AUTH in the phone instead. I wanted to
be able to set up so that each line is a different peer like below:
sip_.cfg:
AUTH = ; secret
LINE1 =
LINE1_PROXY = 1
Since it's not clear from this thread of conversation, do you need 100
unique DIDs? If you do:
That NPA is owned by Pacbell with the central office: SCRMCA12
I don't know if anyone but Pacbell will have numbers in that NPA.
Since I use them and am happy with the service, you can
I'm having a different problem building this release:
[CC] app_dahdiras.c - app_dahdiras.o
In file included from app_dahdiras.c:50:
/usr/include/dahdi/user.h:736: error: expected specifier-qualifier-list
before ‘__s32’
/usr/include/dahdi/user.h:939: error: expected specifier-qualifier-list
. I'm struggling a bit with the relationship
between the user registrations in the phone admin and the lines/users in
the sip_.cfg and sip.conf.
M Hulber wrote:
I have a new Polycom Spectralink 8002 and am having trouble with the
configuration or the unit but I can't see what's wrong
On my other Aastra phone there is a config in the phone menu to specify
the name of the firmware file under Advanced Settings / Firmware
Update. If you have access to that you might see what name it's looking
for. I suppose you would be able to set this name in the config files
also but I
release.
Thanks!
Leif Madsen.
M Hulber wrote:
I'm having a different problem building this release:
[CC] app_dahdiras.c - app_dahdiras.o
In file included from app_dahdiras.c:50:
/usr/include/dahdi/user.h:736: error: expected specifier-qualifier-list
before ‘__s32’
/usr/include
I have a new Polycom Spectralink 8002 and am having trouble with the
configuration or the unit but I can't see what's wrong. The unit does
not seem to even attempt to register with the Asterisk proxy but I can
make calls to it. I have viewed the syslog from the device which it
will actually
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