Hi Max,
In CDR configuration for Pstgres Database, I have copied
cdr_pgsql.so module file from other Asterisk System. May be that happen
in this my problem I have Installed Asterisk 1.4 on CentOS-5 (64bit).
So, exactly what in both system which I have made recently the file size
Hello,
i have one question regarding incoming SIP INVITES.
I have a testbed where i have 5 extnsions : 6001 - 6005
Domain : domainA.com
Then i have configured a sip trunk, where my PBX registers to a foreign SIP
Proxy.
All is working fine, until following scenario:
Incoming call from [EMAIL
I have made class for MOH uploaded a mp3 file to the folder.
Now I am using this class for music on hold during dialing.
Now when call has been established, I put the other end on hold.
So from that end I should listen uploaded file.
But I am not getting audio.
From memory, you need to
Have you looked at PRI-BRI Fail-over-Switches ?
2008/8/27 Jeremy Mann [EMAIL PROTECTED]
We've done the asterisk passthrough route, but if the asterisk box is down
for whatever reason both systems are down.
Splitter wasn't the right word, but yes I see your point, I'll look into
the Adtran.
Doug Lytle wrote:
This is what we do. Along with an ADIT 600 (eBay special)
Just as a side note, I got very lucky on eBay and got my Adit and FXO 8
card for $50 each. I was tickled pink!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Anthony Francis [EMAIL PROTECTED] writes:
Lets not forget that the DECT specification does allow for data
transmission. THere is no reason that in the future you would not be
able to have integrated services over DECT.
The DECT data rate is way too low for integrated services.
/Benny
The DECT data rate is way too low for integrated services.
Does anyone know the actual data rate on DECT?
I've never built any wireless mobility apps, but I would assume the
bitrate required would be quite low, being mostly XML text. In
comparison to the audio data stream it would seem
While I myself am not a MagicJack user, I'm curious as to whether anyone
here has managed to heist their MagicJack account's sip credentials, and
use them to terminate calls using asterisk. Apparently it's as simple
as sniffing the SIP credentials. If so, said person would enjoy
unlimited
Have you looked at PRI-BRI Fail-over-Switches ?
We've done the asterisk pass-through route, but if the asterisk box is down
for whatever reason both systems are down.
Look at this brand new failover device:
http://www.rhinoequipment.com/1portfail.html
Has anyone ever really, truly, actually held on to a Wi-SIP call while
moving from the range of one AP to the range of another AP in the same
network?
Let's say a 'YES' only counts if you had a bona-fide handoff. In other
words, you began in place 'A' (within range of AP#1 but OUTSIDE the
On Sat, 30 Aug 2008 11:51:49 -0500, Karl Fife wrote:
Has anyone ever really, truly, actually held on to a Wi-SIP call while
moving from the range of one AP to the range of another AP in the same
network?
Let's say a 'YES' only counts if you had a bona-fide handoff. In other
words, you began
On Saturday 30 August 2008 11:51:49 am Karl Fife wrote:
Let's say a 'YES' only counts if you had a bona-fide handoff. In other
words, you began in place 'A' (within range of AP#1 but OUTSIDE the
range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but
completely outside the range of
On Sat, 30 Aug 2008 09:05:58 -0500, Karl Fife wrote:
The DECT data rate is way too low for integrated services.
Does anyone know the actual data rate on DECT?
I've never built any wireless mobility apps, but I would assume the
bitrate required would be quite low, being mostly XML text.
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
-- Executing Dial(SIP/2000-081b9938, Zap/g0/0501125||) in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0501125
On Sat, 30 Aug 2008 11:31:26 -0500, Karl Fife wrote:
While I myself am not a MagicJack user, I'm curious as to whether anyone
here has managed to heist their MagicJack account's sip credentials, and
use them to terminate calls using asterisk. Apparently it's as simple
as sniffing the SIP
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten
2008/8/30 Shariq Khan [EMAIL PROTECTED]
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
-- Executing Dial(SIP/2000-081b9938, Zap/g0/0501125||) in new
stack
-- Requested transfer
wouldn't the ap ranges have to have *some* overlap, lest the basic
network
connection be dropped, whereby dropping the voip call?
Indeed you're right.
You'd have area covered by AP 'A' only, AP 'B' only and area of AB
overlap, Picture a venn diagram:
sean darcy wrote:
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first
On Saturday 30 August 2008 01:35:10 pm Karl Fife wrote:
Indeed you're right.
You'd have area covered by AP 'A' only, AP 'B' only and area of AB
overlap, Picture a venn diagram:
http://upload.wikimedia.org/wikipedia/commons/5/56/Venn-diagram-AB.png
right. it's just your inital description
Krunal Patel wrote:
Hi,
I have a simple dialplan.
[test]
exten = _X.,1,Dial(SIP/1000SIP/1002)
I have registered user whose context is test.
Now I am dialing any number, so it will enter into test context.
It will dial 1000 1002 both.
Both keeps ringing.
Now the problem is, when any
The Siemens DECT line are open source. Not broadly available in the US
though.
What does this mean? Could you please provide a link for more
information?
here's a link to the Siemens open source dect/wifi phones if that's
what you are looking for.
Michael Graves [EMAIL PROTECTED] writes:
DECT was designed from the start to handle voice and data.
553kbps isn't particularly useful today. Even 3G is faster.
Not that I have ever seen products offering more than ISDN speeds, and
I believe that was before 2000. DECT is excellent, but it
What did you try and how did it fail? Are you using the t option in queue?
On 8/30/08, Mark Hamilton [EMAIL PROTECTED] wrote:
So, no answers or is this thread going to remain unanswered too?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: August 28, 2008
last time i had this issue with teliax, they recommended to upgrade to 1.4
On Fri, Aug 29, 2008 at 3:44 AM, Chris Mason [EMAIL PROTECTED] wrote:
I tried DTMFmode=auto and it did not help. Any further ideas?
--
This message has been scanned for viruses and
dangerous content by MailScanner,
Asterisk Users -
We are presently try to operate a hybrid GSM/Asterisk cellular
basestation at the Burning Man Festival in the Nevada desert. (See
http://openbts.sourceforge.net). The architecture is basically one
where cell phones are presented to Asterisk as SIP users, using the
IMSI
MagicJack has back hacked for some time
http://revolution.hackthisbox.com/magicjack/readme
Cory J. Andrews
Director New Market Initiatives
VoIP Supply, LLC.
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
[EMAIL PROTECTED]
Have I exceeded your
On Sat, 30 Aug 2008 23:41:16 +0200, Benny Amorsen wrote:
Michael Graves [EMAIL PROTECTED] writes:
DECT was designed from the start to handle voice and data.
553kbps isn't particularly useful today. Even 3G is faster.
Not that I have ever seen products offering more than ISDN speeds, and
I
On Saturday 30 August 2008 19:15:36 David Burgess wrote:
Now we've discovered a new problem: Asterisk lets these non-existent
make calls even though they are not listed as users in sip.conf. We
suspect that is happening because they are all localhost connections,
and therefore bypassing some
I have a DID with budgetphone.nl, which has worked fine for quite some
time. For the last few weeks, most (but not all of the time), the incoming
call does not go where it is supposed to, but instead the following message
show on the console:
[Aug 30 19:47:28] NOTICE[5161]: chan_sip.c:13865
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