Displaying 20 results from an estimated 40000 matches similar to: "SPA3000 + asterisk +call waiting"
2008 Jan 25
2
SPA3000 -- PSTN to VoIP
Hi, all
I am trying to figure out how to forward incoming PSTN call on SPA3000
to VoIP extension(s).
Basically, I have converted my home to VoIP. I have normal phone
(connected to SPA3000) and couple of IP phones. All call coming from
VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I
need to do same thing for incoming PSTN calls.
I have enabled gateway function in SPA3000 and
2004 Jun 29
0
MGCP and call waiting, doesn't work.
Hey guys, can you shead some light on this?
I will copy my mgcp.conf and post below, but here is the problem.
I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick
2006 Mar 10
0
Flash call transfer problem
Hi,
I have some problems transfering call from phone to phone with my Asterisk. When I dial Flash I can hear for half a second the dial tone, but it stops suddenly. The other phone hear the on hold music and pressing flash key another time I get back to the previous channel.
On the asterisk consolle seems to be all ok, this is whant I can read:
asterisk1*CLI>
-- Swapping 0 for 1 on
2006 Dec 14
4
Zaptel under FC6
Hi, all
I am building a new server. Have installed FC 6 and put in TDM400 card.
Checked out latest asteriusk code, run make install in zaptel directory.
So far all is fine.
Now I am trying to install the drivers.
# modprobe zaptel
FATAL: Module zaptel not found.
Fair enough, no zaptel driver is found on the system.
Is there are any known problems with FC6? I did not have much trouble
running
2006 Mar 13
2
DISA & SPA3000 issues
Hi,
These days I run into something quite odd.
I have an A@H that was modified to meet our requirements.
We have a completely funtional DISA which we use pretty much all the
time.
I works flawlessly with incomming SIP calls from several providers,
IAX calls from FWD and with ZAP.
Recently we came out with a situation where it doesn't work... with
a
2007 Jan 23
12
How to exit from console?
Hi, all
Stupid question, but how do you exit asterisk console without stopping
the asterisk?
Tried quit and exit:
*CLI> exit
No such command 'exit' (type 'help' for help)
*CLI> quit
No such command 'quit' (type 'help' for help)
*CLI>
Any other ideas?
I started asterisk with -cvvvvg option. Same problem if use asterisk
-r to connect. Can not exit.
Any
2006 Aug 12
1
SPA3000 dialplan coding...
Hi all,
Can anybody explain what these values exactly mean. As you all know its the
dialtone value on an SPA3000 of linksys.
350@-19,440@-19;10(*/0/1+2).
Can anybody help me how to write this code for a dialtone of frequency 425
which is continous.
Thanks
Dan
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2006 Feb 20
2
spa3000
I'm trying to get working a spa3000 with asterisk. My problem is I
cant get wroking the incomming calls (I installed the lastest
firmware). My problem is asterisk is rejecting the authentication from
the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I
think I placed the username and password correctly...
Sip.conf says this:
[linea2]
username=linea2
type=peer
secret=1111
2005 Feb 25
1
Seting up for afirst time -- can not call
Hi, all
I am setting up Asterisk for the first time and have some problems.
Setup is very simple -- Astersik box and two Polycom SP300 phones. I will
add bells and whistles as I go, at the moment things are very simple. No
TFTP servers, so phones run with their default configuration.
I set up IP addresses, netmask and gateway IPs manually on the phones.
Now, I have read of problems with
2005 Sep 10
2
VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port 11112
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is
2006 Mar 02
3
Sipura SPA-3000 vs Linksys SPA3000
Hallo!
I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out
to be unreliable and never shipped.
Yesterday I went looking for alternative suppliers and found the Linksys
SPA3000 device. It's a different box, but the specs look very similar.
Is this the same device? Has anyone used this Linksys SPA3000
successfully with Asterisk?
Thanks,
Frank
2004 May 11
1
Sipura, Asterisk, *0, and Call Waiting
I have searched through all the lists and found that some people have had
luck with flash hook, then *0 to answer a call waiting call.
I have an Asterisk server with one FXO card, the dialtone for the fxo card
is providing by another pbx called a Definity.
When I am on the Sipura, and another call comes in, I hear the call-waiting
indicator, when I flash hook I just hear tone, if I dial *0 I
2003 May 16
4
How to handle call waiting?
Hello All,
I need to be able to pass hook flash from an extension on a TDM400P to the
analog line on an X100P FXS to use telco-provided call waiting feature. I
know, i know, this is evil and I need to get more lines in a call group, but
I dont think this is very appropriate for a home answering machine or a
one-persone office.
In any way, I do have callwaiting=yes in the zapata.conf. When I
2005 Aug 22
0
SPA3000 dial plan?
Hey, all... If this is too off-topic, I'd be grateful for directions to a
more appropriate mailing list.
I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000
which is registering twice with Asterisk - once for its FXS/Line1/VoIP1
and once for its FXO/PSTN/VoIP2.
My eventual goal is to have inbound calls on its FXO ring four times on
its FXS and then fail over to
2003 Oct 20
1
mgcp transfer takeback with ata186 (logs with comments - long post)
Hi,
in following of a recent discussion I got to work on MGCP with the Cisco
ATA186 again, and got it to work very nicely. However, there is a little
thing with transfers I would like to get comments on:
Call comes in from PSTN and goes to an ATA186 (MGCP)
Call is answered and then, using flash, transferred to another extension
If the extension is available, there can be an announcement and
2003 Oct 22
0
MGCP error for Cisco 7750 FXO card
Can anyone tell me what MGCP error that I'm getting means?
The hardware is a MRP200 in a Cisco 7750 PBX. (Its a FXO blade with 2 slots, first one has a 4 port FXO card and the second has 2 port FXO card. It recognises those correctly, at least to the point of this error.)
MGCP Debugging Enabled
MGCP read:
NTFY 13 aaln/S0/SU0/0@MRP200-S1 MGCP 0.1
X: 1adace42
O: L/hd
from
2007 Jul 11
2
Call Waiting
Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting.
Using a TDM400p, with FXO and FXS modules.
On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a "hook flash", to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can
2010 Jan 25
0
[OT] spa3000 (Regional & Line1) NL settings required
Dear all,
Can someone from Netherlands who has SPA3000 send me the Regional & Line1
settings.
I'm not sure what is wrong but I can call from asterisk to phone attached to
spa300 FXO, but not the other way. I tested three phones: siemens gigaset,
tiptel 160 and hpoj k80(fax). Only tiptel 160 can call to the asterisk,
other two just dial but nothing happens. I guess this has something with
2003 May 19
1
MGCP and Cisco ubr924
I've been trying to figure this one out for a while, but to no avail.
I have my cisco ubr924 setup for MGCP with Asterisk as the call-agent. I have manually registered the endpoint in mgcp.conf. When I pick up the phone, I get no dialtone and debug shows errors. IOS on the ubr924 is 12.2.
Any help is appreciated.
from mgcp.conf:
[ubr924]
host=65.37.86.203
context = from-sip (just as a
2003 Jul 11
3
mgcp problems
I strange error messages when using mgcp and ata186 .
This session is simply dial into 600 demo extension - echo test
...
Handling request 'NTFY' on aaln/1@10.0.1.19
Transmitting:
200 29 OK
to 10.0.1.19:2427
-- Endpoint 'aaln/1@10.0.1.19-1' observed '0'
-- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode:
sendrecv
Posting Request:
RQNT 306