Displaying 20 results from an estimated 10000 matches similar to: "OT - SPA3102 not detecting CID - Which settings to tune ?"
2007 Oct 22
1
[France CID] Does Zaptel support ETSI FSK?
Hello
I've been googling for a couple of days now, but still can't
figure out what to put in zapata.conf to get it to report CID.
Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202
as CID FSK Standard:
http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg
http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg
Does Zaptel support those on Digium
2008 Mar 05
1
Linksys SPA devices and CID
Hi list,
After successfully configuring Linksys SPA3000 and SPA3102 devices as
Asterisk PSTN gateways, the only thing I can't get working is the PSTN
Caller ID. The analog and SIP phones I've used can both display CIDs
for internal calls, while the analog model also displays CIDs
correctly when attached directly to the PSTN line. However, when PSTN
calls come in via the SPA
2008 Feb 18
2
SPA-3000 caller ID and KPN
Hi list,
Hopefully, some of our Dutch members can help with this one. I'm also
based in the Netherlands and am using a Sipura (Linksys) SPA-3000
(firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test
system. It works fine, except that the Called ID (CID) is not working.
I'm aware that KPN (our local telco) requires a separate subscription
to activate CID on POTS
2007 Oct 23
0
Internal Data Stream Error
Hello again,
I am using mix monitor and the majority of the sound records perfectly.
I then get a "Internal Data Stream Error" near the end of the sound
file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs
and an example dialplan entry is ;
; phone line phone1
exten => phone1,1,Answer()
exten => phone1,2,MixMonitor(test.wav|av(0)V(0))
exten =>
2005 Oct 12
0
X100P callerid ETSI - caller*ID failed checksum
Dear All,
I am a newbie about asterisk. I have 1x X100P card 3x Sip phone
I got aware of problem, after I saw the caller id on my sip phone. I
noticed that if I receive a call from GSM Operator A, I can see caller id.
But any other operator, I got no caller id, even my direct PSTN service
operator. So at that moment I was using *1.0.9. than I changed to
asterisk@home 1.3(1.0.9). I got same
2004 Apr 14
1
CND (CID) woe on a TDM400P
G'day list,
It appears I got ahead of myself in believing I'd solved my CND (caller
id) problem. My callerid.c change got it working on one of the handsets I
have, but not the other. (Recap: The AU CND standard has an increased
delay between the end of the ring signal and the start of the CND FSK data
-- it's 500ms in US GR-30-CORE, and around 700ms in AU DC.002/TS030).
I've
2010 Mar 01
0
SPA3102 Firmware Upgrade via TFTP fails
Hi everyone,
I'm trying to set up a VOIP mass deployment.
To do so, I want to generate a configuration xml fails.
I read somewhere that I had to use : http://<phone -or- device ip
address>/admin/spacfg.xml
but it work with an upper firmware only.
My Software Version is 3.3.6(GW)
The last firmware versin is 5.1.10(GW)
Upgrade Enable: is set to yes on the linksys SPA3102 web interface
2012 Jun 03
2
Caller ID : FSK ETSI or FSK US
Hello, All :)
Regarding to incoming caller ID on PSTN line, which one is best supported
by asterisk: is it FSK ETSI or FSK US?
I bought some caller ID converter hardware (convert DTMF to FSK and vice
versa) but still asterisk can not detect it.
The converter has a switch FSK ETSI or FSK US
This is what I put in /etc/asterisk/chan_dahdi.conf
...
cidsignalling=bell
cidstart=ring
...
If after
2010 Dec 06
1
[3102] How to rewrite CID name + number?
Hello
I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:
http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png
The problem is that by default, Asterisk doesn't rewrite the CID name
+ number in incoming calls, so that XLite displays whatever name I
used in the 3102 and the extension the 3102 uses to register with
Asterisk.
How can I tell
2009 Mar 17
3
SPA3102 - How to save config in a file
Hi,
I've read in this mailinglist archives some notes related to Linksys SPA3102
provisioning but I couldn't find there the answer I'm looking for.
Is it possible with this box (mine is unlocked) to store its config file(s)
in a TFTP server, and have this(these) file(s) reloaded at boot time, for
instance ?
In embedded web server, there is a Provisioning tab full of settings but
none
2010 Mar 19
0
SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )
Ok,
I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is
established asterisk seems to drop the call.
However I still hearing ringback on pstn side, call is established again,
and asterisk drops the call again, like a loop.
-- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948",
"horario-atencion/our-business-hours-are") in new stack
2012 Mar 10
1
SPA3102 asterisk signaling
Hy all,
Recently a have a little problem with a Cisco device, SPA3102. I use
this device with asterisk to dial out with outbound trunk. (SPA3102
has 1 FXO port)
It working ok , but the device SPA3102 do this : when a call is placed
for outgoing in asterisk and send to SPA3102 , this device "answer
and dial the number in the same time" , in my CLI I see the channel
is open , but on
2013 Nov 05
1
How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?
Hello,
I've got an analog phone which is currently receiving unsollicited FAX
calls from PSTN.
For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would
let voice calls come in and out and translate incoming FAX calls to TIF
files (forwarded through email)).
My target setup is :
PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->
2008 Feb 27
1
SPA3102 registration problem
Hi list,
After failing to get a Sipura/Linksys SPA3000, which I've configured
as a PSTN gateway, to pass on the Caller ID, I decided to try my luck
with a Linksys SPA3102 after hearing some promising stories.
Unfortunately, I've run into a completely new problem: it seems
Asterisk won't let this device register.
I went about configuring the SPA3102 in much the same way as I
2007 Jul 30
1
Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
Hi All,
In our small office calls to the PSTN are currently sent via Asterisk and a
Linksys SPA3102 (1 x FXO and 1 x FXS):
SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN
If the PSTN is in use on SPA3102 I need a way to get the call to then route
out over IAX termination.
SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)
2007 Dec 30
2
asterisk callerid
I'm missing something simple I think:
I have an spa3102 for which I want asterisk to use the incoming pstn
callerid when it sends the call to a local extension (207).
callerid works fine for the internal phones (between each other)
The spa3102 is picking up the PSTN callerid and displays it in its own
status pages
Asterisk however, doesnt see the callerid at all.
The spa3102 is set to:
2003 Dec 20
0
X101P + TDM400P
I thought I'd share my Asterisk experience, which hasn't exactly been as
pleasant as I would like but now seems usable in most ways and more then
I expected in other ways. I wanted a home PBX system, that would let me
treat different callers different ways depending on CID.
I initially bought the Digium developer's kit to try things out. That's a
single port TDM400 and a X101P.
2007 May 08
1
Problems witch SPA3102.
Hello,
i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with
cdr. Well all I want is to receive incoming calls from pstn on specified sip
account (suppose 8000), and to initiate outgoing calls from all my asterisk
sip accounts through SPA3102 device. Someone can explain me what may i set
on SPA and asterisk to do this thing. Thank you for your support.
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2010 Mar 18
1
SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)
Somebody has 5.1.7 firmware for SPA3102?
I'm having issues with inbound/outbound calls using asterisk through SPA3102
with firmware 5.1.10. I've read it has a codec bug, since it doesn't care
about what you set up in Preferred Codec.
Any help will be appreciated.
Sebastian
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2008 Oct 16
0
Sharing my Asterisk + SPA3102/PAP2 setup: What I've learned in 1 week.
(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified
answers become).
Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario:
> Hey Rodolfo... Need some help from you ...
> I need to know what hardware do I need to make SIP calls if I set-up
> asterisk
> So the situation is that I have a PC and configure the software of my PC to