Displaying 20 results from an estimated 3000 matches similar to: "Sip Transfer"
2003 Mar 27
9
Dlink DG-104S
Does anyone know if this unit works with Asterisk?
Thx.
B.
2003 Aug 10
2
SNOM200 firmware roll back!!
Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q..
Anyone know why?
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2006 Nov 17
11
wget from within asterisk?
What would be the simplest way to retrieve information form a CNAM
database that provides http based query responses?
Does an application or script already exist that does this?
Basically, I want to do a wget of a URL that contains the callerID
number as a variable, and assign the returned text to another variable
which can be used to set the caller ID name.
Any suggestions?
2003 Aug 07
1
Sip Trunk config
incominglimit is already implemented for SIP. Just specify under the
endpoint how many incoming connections are allowed.
For example,
[cisco]
type=friend
username=cisco
secret=blah
nat=yes ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms away
2003 May 17
3
E400P and 2 X100P working, but not together
Hey all,
?
I'm trying to get an E400P and 2 X100Ps working together in the one box
and don't seem to be having much luck.
?
I can get the two different types of boards working separately, but not
together.? I've made calls on both the X100P and have seen sync and
correct signalling on the E400P.? But when I try to enable to configs
together I get the following:
?
modprobe wcfxo
2003 Sep 11
1
Incoming calls from IAXTEL over NAT
Hey all,
I was playing around with IAXTEL last nite and have
outgoing calls working a treat. I'm sure I woke a few
people up in the US with my annoying test calls. :)
Anywayz, incoming calls are a different matter. I have a
NAT firewall my * box is sitting behind and the server
'appears' to have registered correctly with IAXTEL. Thing
is, when I try and call my 1700 number
2004 Dec 14
3
Confirm MWI doesnt work with SIP RealTime?
Can someone else confirm that your phone does not recieve MWIs when using
SIP and RealTime?
Is this a problem with SIP or with Voicemail?
-Matthew
2003 May 26
1
Intel 536EP Full Duplex support
Hi
Anybody have a example or use this modem as FXO, acording to intel this
modem is a real full duplex modem, so i have one question why i can we
use this modem as fxo
My idea is simple when you recive a call in the modem you can transfer
this call to any other place using for example a IAX2 or Openh323
protocol
This is posible?
Thanks
2003 Sep 25
15
CDR Web Search Frontend
*This message was transferred with a trial version of CommuniGate(tm) Pro*
Hey all,
I've just done a quick (but functional) web front end for searching the
CDRs in a MySQL database. Anyone interested in trying it out? I'm
wondering what to add to it next.
So far you can seach using source, destination, CLI, channel and date
ranges. It also displays ALL fields in the database table.
2003 Aug 12
1
usrobotics modem and pstn
hi,
i have a external usrobotics modem, i want to use it with asterisk to
interact with the pstn,
what i have to do?
thanks,
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universidad del cauca
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2003 Aug 17
2
no incoming packets & Sound: Recording overrun
On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote:
> Hello, and thank you for registering at gnophone.com. Your login
> information is listed below:
>
> Username: miernik
> Password: *******
> IAX Phone Number: 17002916107
>
> Please login as soon as possible to
> http://x.linux-support.net/directory/ to complete the
2003 Aug 14
2
Don't know how to calculate timelen
Hi all,
I'm setting up my first * install and have it peering with another * machine
using IAX across the internet which provides our pstn gateway.
So far I have the IAX "friend" set up correctly but when I make a test call
from an external phone, I get:
WARNING[5126]: File chan_iax.c, Line 648 (get_timelen): Don't know how to
calculate timelen on 8 packets
I have set up a
2003 Nov 05
1
iconnect
Hi,
I was able to connect asterisk to iconnect's service.
It took me almost two hours, but it's because I was having NAT trouble.
I finally discovered that you can set the iconnect host to
natrealy.deltathree.com to make it work.
(for those of you who, like me, don't have the time to search the
archive I'll provide a working sample in a minute)
My problem was sound
2003 Aug 06
3
X-Lite <-> Snom200
Hi,
I have just been playing with the latest X-Lite.. It works fine with Asterisk..
As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why..
But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that
2003 May 26
9
The Phantom Call..
My system seems to be generating a call on its own... Unfortuately I can't give much more information..
I have an X100P and an S100U..
My Modem and the X100P share a common line.. When I am on the internet (which is most of the day) * just sits there and does nothing (apart from when I am testing ideas for the dial plan), but at night when I am sleeping and the modem is not connected then
2003 Aug 12
2
How to Asterisk
Hello,
I'm new user of asterisk. Can anybody pls tell me how to use asterisk or any detail how to link????
i installed Asterisk-0.4.0 on i810 onboard sound card with Redhat 7.1.
when i type "asterisk -vvvc" i get *CLI> prompt
Prakash
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2003 Dec 12
1
simple question on sip.conf
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal or another SIP proxy can
redirect his calls to my * as well. Those calls two will come through
general
2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2003 Mar 03
3
iconnecthere 480 error: is there a workaround?
I am going to have to find a fix for this problem or I'm going to have
to quit using iconnect.
About one call in 10 or so, iconnect's gateway gives me an error
(console output appended below).
So upon receiving the error, which as a 4XX error means, "Fatal,"
asterisk gives up and drops the call. But not iconnect!! The phone at
the other end starts ringing, and rings
2003 Aug 19
5
SIP QUESTION
Hi
Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C
Site A Site B Site C
ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186
Thanks
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