Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash
I've managed to get a basic system set up. and can now take and make
sip calls over the sip trunk I've got from sipgate.co.uk for testing
purposes
Anyway I can make calls fine (if only to the testing line and other
sipgate lines
do you use the
qualify=yes
option for your endpoints?
y.
Peter Childs schrieb:
Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash
I've managed to get a basic system set up. and can now take and make
sip calls over the sip trunk I've got from sipgate.co.uk for testing
2010/1/26 Yves Arikoglu yves...@gmx.de:
do you use the
qualify=yes
No, If I do it does not work at all.
I've found if I set defaultexpiry to 30 it works fine. and was infact
working for 30 seconds every two minutes before, It looks like
sipgate.co.uk are expiring there registry attempts very
Hi, does anyone have an info into what could cause
[Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken
pipe
[Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken
pipe
[Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken
pipe
[Nov 28 14:26:23]
2010/1/26 Peter Childs pchi...@bcs.org:
2010/1/26 Yves Arikoglu yves...@gmx.de:
do you use the
qualify=yes
No, If I do it does not work at all.
I've found if I set defaultexpiry to 30 it works fine. and was infact
working for 30 seconds every two minutes before, It looks like
I have always experienced very good call quality via my asterisk
server snom phones.
With soft phones especially on mobile (sipdroid or asip) quality
is poor. With commercial sip services on my mobile - that do not
of course use my asterisk server but use the same good wifi
connection - quality
Kingsley Tart wrote:
Thanks for the link. I looked at that page but couldn't see how it
helped with my specific issue, unfortunately, though I admit I'm fairly
new to asterisk so I don't fully understand what's going on.
The page gave an indication, as Lee stated, that you'd see this
AIR, the $GARBAGE sort of forces a match. The way I test a
grammar/background is something like this:
(this is longer because it uses Waitexten when no lumenvox license is
available)
[main-menu-select]
exten = s,1(start_menu),noop(select 0-9 except 8)
exten = s,n,Gotoif($[${usedtmf} =
John Millican wrote:
Hello,
I have a situation where a remote worker dials in to the asterisk server,
enters
the secret code, then dials out via Disa on a PRI. This was all working
great
until this morning. Now the calls is placed out, connected but there is no
voice from/to either
Hi All,
i want to make an extension from pbx1 able to tlak to another extension from
pbx2 or use pbx2's trunk to dial outside calls.
so i edited in both servers accordinally the iax.conf:
register = pbx1:p...@172.16.200.175 pbx1%3ap...@172.16.200.175
[pbx2]
type=friend
host=dynamic
trunk=yes
The idea is that the Queue() application uses different strategies to ring
agents, so it decouples you from having to worry about that. You could have
that by setting the queue to rinagll strategy.
l.
2010/1/25 bhrugu mehta mehtabhr...@gmail.com
Hi, all
Is ther any way to pass channel queue
This is how I did it..
I have to Servers, SRV1 and SRV2
In SRV1 iax.conf
[SRV1-SRV2]
type=peer
username=SRV1-SRV2
secret=Password1
host=IP OF SRV2
qualify=yes
[SRV2-SRV1]
type=user
username=SRV2-SRV1
secret=Password2
context=from-iax
host=IP OF SRV2
quailfy=yes
If
I realize that many of you are too far away to consider it, but I know
of a couple of people who are considering going. Is anyone tempted? I
am planning on going and have a promo code for you if you'd like one.
r
--
_
--
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 figure out if it is possible to somehow get a
transferred call back to the transferrer
Johann:
Do you know how is the SMS sent over the IP, does it use SIP INFO
message or somthing like that?
Regards,
Juan
Johann Steinwendtner wrote:
randulo schrieb:
2009/10/9 "Juan E. Rodrguez" jerdg...@gmail.com:
Does any one know about a SIP hard phone capable
Hello,
I have found a number of postings about XXX Message longer than it
should be?? XXX but I guess these problems have been fixed in the
current versions.
obviously this was not true.
In the file q931.c of libpri-1.4.20.2 I had to remove line 3458:
return -1;
After recompiling
When I run make install I don't see this file getting overwritten. Do
I have to delete it to get this to happen?
On 1/25/2010 7:06 PM, Tilghman Lesher wrote:
On Monday 25 January 2010 08:52:45 Mark Hulber wrote:
Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had
On Tuesday 26 January 2010 10:08:39 Mark Hulber wrote:
On 1/25/2010 7:06 PM, Tilghman Lesher wrote:
On Monday 25 January 2010 08:52:45 Mark Hulber wrote:
Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had
this problem before and even when I move to back versions I have
I've found that I get this error when I don't properly listen for
asterisk responses to my commands in my agi scripts. Anytime you send
a command to asterisk from an agi script, asterisk sends a response to
the script with the result of the command (i.e a 200 ok response if
asterisk was
On Tue, 2010-01-26 at 07:46 -0500, Doug Lytle wrote:
Kingsley Tart wrote:
Thanks for the link. I looked at that page but couldn't see how it
helped with my specific issue, unfortunately, though I admit I'm fairly
new to asterisk so I don't fully understand what's going on.
The
Kingsley-
On Tue, 2010-01-26 at 07:46 -0500, Doug Lytle wrote:
Kingsley Tart wrote:
Thanks for the link. I looked at that page but couldn't see how it
helped with my specific issue, unfortunately, though I admit I'm fairly
new to asterisk so I don't fully understand what's going on.
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
Jeff Brower wrote:
How do you know for sure fax detection is turned off? It sounds to me like
your changes to the dahdi config file are
being ignored. Maybe put something in there that should cause an error or
something clearly observable, then see
whether that actually occurs.
Or even
26 jan 2010 kl. 16.48 skrev Örn Arnarson:
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 figure out if it is possible to
Great, do you know of any other files outside of
/usr/lib/asterisk/modules that get recreated? I also place
rc.redhat.asterisk as asterisk in /etc/rc.d/init.d I don't see that
safe_asterisk_restart gets placed anywhere. It looks like astgenkey and
autosupport both get written over.
On
On Fri, Jan 22, 2010 at 7:50 AM, Mike l...@virtutel.ca wrote:
I know having Asterisk aware of Polycom Do No Disturb state wasn't working
before (1.4), but is this working in any recent version? Is there any
custom way of doing this?
Our Asterisk servers (1.2 and 1.4) get SIP response 603
hi,all
i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except
realtime queue.
it seems queue_table works fine, but queue_member_queue not work, the
two tables works fine when in 1.4.28.
is that something changed related to realtime queue configuration?
more detail about two
I am using 1.4.21.2 and DND is definitely working.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Saturday, 23 January 2010 2:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial
On Mon, Jan 25, 2010 at 12:07:55PM -0600, Karl Fife wrote:
From: cb c...@mythtech.net Sent: Sunday, January 24, 2010 12:42
I use the Snom 370 all day long at work. I have never had a problem
adjusting the volume. I change it multiple times a day as I keep my
handset on one volume and my
Hi, does anyone have an info into what could cause
[Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe
I have had the same issue with a PHP script that logs into the manager
interface. If you don't wait for the AMI response, then log off before closing
the connection
2010/1/26 Tilghman Lesher tles...@digium.com:
On Monday 25 January 2010 03:12:08 Zhang Shukun wrote:
hi, dear all
MYSQL commands work well in 1.4.28 edition, but not in 1.6.21
is that the grammar is different between them?
extensions.conf
exten = s,2,MYSQL(Query resultid ${connid}
2010/1/23 Steve Edwards asterisk@sedwards.com:
On Fri, 22 Jan 2010, Zhang Shukun wrote:
as you know, we can use MYSQL command to visit mysql database
but if i use other database like Oracke,sybase,etc, Could i use MYSQL
command ?
ODBC will do what you want.
Thanks, while i think
2010/1/26 Carlos Chavez cur...@telecomabmex.com:
You must read the upgrade instructions. The database definitions in
res_mysql.conf have changed. The way you reference the database in
extconfig.conf is also different.
solved...
it is my configuration error of res_mysql.conf and
2010/1/22 Tilghman Lesher tles...@digium.com:
On Friday 22 January 2010 04:06:29 Zhang Shukun wrote:
2010/1/22 Randy R randulo2...@gmail.com:
On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun bit...@gmail.com wrote:
exten = 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
but what should i
Un-mid-posting...
On Fri, 22 Jan 2010, Zhang Shukun wrote:
as you know, we can use MYSQL command to visit mysql database but if i
use other database like Oracke,sybase,etc, Could i use MYSQL command ?
2010/1/23 Steve Edwards asterisk@sedwards.com:
ODBC will do what you want.
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