Connecting PSTN line to SPA112 2-Port Phone Adapter

Hi,
Not sure if this question is right but can I use the SPA112 2-Port Phone Adapter to connect a PSTN line? What I want is, I should be able to choose a line and while making calls. Use PSTN or VOIP.
Please let me know if I am using a wrong device for my purpose.
Regards,
Naidu

No, it cannot, that device has FXS ports, no FXO ports
Includes two standard telephone ports, each with an independent phone number, for use with fax machines or analog phone devices
http://www.cisco.com/c/en/us/products/collateral/unified-communications/small-business-voice-gateways-ata/datasheet_C78-691106.html

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Connecting FAX to PSTN line using FXS and FXO ports

I have 2 cisco 1760 routers with FXS and FXO card installed in each. I have to transport 2 PSTN lines from Head Office to remote loaction using FXS and FXO cards and these lines will be used for voice calls and FAX( one line each for FAX and voice). I have configured the router for voice lines and it is working fine. Now I am using the same config for running fax machine but it is not working. Can anyone help me out how to configure FAX in this scenario also if anyone can share any sample config. I am attaching my config for both routers. Right now both routers are connected with a cross-over cable for lab test but we will connect them later using satellite connection.
HO Router (FXO card)
Current configuration : 1718 bytes
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
no aaa new-model
resource policy
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
voice-card 3
ip subnet-zero
ip cef
no ip dhcp use vrf connected
no ip domain lookup
no ftp-server write-enable
voice class codec 10
interface FastEthernet0/0
ip address 10.10.10.1 255.255.255.248
speed auto
h323-gateway voip bind srcaddr 10.10.10.1
ip classless
no ip http server
control-plane
voice-port 2/0
output attenuation 0
echo-cancel coverage 32
no vad
no comfort-noise
timeouts interdigit 3
timeouts call-disconnect 3
connection plar opx 2001
description Remote PSTN#:35296913
music-threshold -70
voice-port 2/1
output attenuation 0
echo-cancel coverage 32
no vad
no comfort-noise
timeouts interdigit 3
timeouts call-disconnect 3
connection plar opx 2002
description Remote PSTN#:35296914
music-threshold -70
voice-port 3/0
voice-port 3/1
dial-peer voice 2000 voip
destination-pattern 200.
no modem passthrough
voice-class codec 10
session target ipv4:10.10.10.2
incoming called-number .
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 7200
fax nsf 000000
no vad
dial-peer voice 1321 pots
description line 1
huntstop
destination-pattern 1321
port 2/0
dial-peer voice 1322 pots
description line 2
huntstop
destination-pattern 1322
port 2/1
line con 0
password cisco
line aux 0
line vty 0 4
password cisco
login
end
Remote Router (FXS Card):
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
voice-card 3
no aaa new-model
ip subnet-zero
ip cef
no ip domain lookup
no ftp-server write-enable
voice class codec 10
interface FastEthernet0/0
ip address 10.10.10.2 255.255.255.248
speed auto
h323-gateway voip bind srcaddr 10.10.10.2
interface Ethernet1/0
no ip address
shutdown
half-duplex
ip classless
no ip http server
control-plane
voice-port 2/0
description PSTN#:
voice-port 2/1
description PSTN#:
voice-port 3/0
voice-port 3/1
dial-peer voice 2001 pots
description Remote
huntstop
destination-pattern 2001
port 2/0
dial-peer voice 2002 pots
description Remote
huntstop
destination-pattern 2002
port 2/1
dial-peer voice 1320 voip
destination-pattern 132.
no modem passthrough
voice-class codec 10
session target ipv4:10.10.10.1
incoming called-number .
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 7200
fax nsf 000000
no vad
line con 0
password cisco
line aux 0
line vty 0 4
password cisco
login
end

In your voice class codec 10 there aren't any codecs declared.
Add G.711 codec in this way:
voice class codec 10
codec preference 1 g711alaw
If the fax communication fails again try to disable T.38 and try fax passthrough mode:
no fax rate
modem passthrough nse codec g711alaw
fax protocol pass-through g711alaw
Regards.

Connecting to an ISDN BRI on a router from a PSTN line from remote site..!!

Friends,
I have a 1812 at HO with WIC1 BS/T Card, connected it to an ISDN BRI line.One of my critical remote location uses Leased circuit to reach my HO Router.But in case if LL goes down,some of the important PC's at the remote site should be able to dial to HO Router BRI Number using PSTN line, get connected and continue, ip address assigned from HO Router and access the application... Is any way available...

Hi Arnab.
You need to configure dial backup through AUX port. You need to connect a normal PSTN line to a modem which in turn is connected to your AUX port. You will have to configure a chat script which you can find in the attachmnet. The attachment is a working example in one of my routers.
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--Pls rate if useful---

Spa3102 would not forward a voip call to pstn line

Good morning.
I've done the implementation provided here http://community.linksys.com/t5/VoIP-Adapters/SPA-3102-and-softphone-to-
make-calls-via-pstn-line/td-p/326390 .
It is a way to use for outgoing calls a given pstn line from anywhere I have internet (voip to pstn).
The spa3102 is connected to a router (with an active DHCP server and ip 192.168.1.1) from where it takes the internal
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LINE) ( but for 5060 for the LINE 1 also). I want to make calls from a voip softphone (x-lite 4) to the SPA 3102 and
this to forward the voip calls to PSTN line to which it is connected. In x-lite the SPA3102 is set as a proxy so that
i can type the phone number I want to call without being followed by the SPA3102's ip each time ( eg on x-lite I
give call number 2101111111 instead of 2101111111 @ wanip: 5061 where wanip is the external ip of the router).
When x-lite is running on the computer that is on the same network with the SPA3102 everything works as expected. A
voip call is made from x-lite ( using as a proxy the wanip everytime, or even for test purposes the dyndns domain
that i set up for this reason), this call is passese PSTN line and the phone of the called party rings . At x-lite
COMES indication "call established ".
The problem occurs when I do the same procedure from x-lite installed on a computer belonging to another network (
e.g. in another building with its own internet connection , own router, own computer , etc. ) . Always using the
wanip the x-lite makes the voip call to the SPA3102, writes "call established" ( meaning it connected to SPA3102) but
never routed the call to the called party ( the SPA3102 did not forward voip calls it receives to the PSTN line ) .
Trying to find what 's wrong I've tried to disable all firewalls (soft and hard from all involved machines ) . The
behavior is the same either the computer that makes the successful calls is connected to the network directly to the
router or through the port "ethernet" on the SPA3102.
What is the difference in these two voip calls to the SPA3102 and the one " triggers " it to forward the call to
PSTN line and the other does not ?
Thanks now for any ideas you give .

The audio sound problem is more than likely also associated with the overall addressing problem initially encountered. As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets. In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets. The sound is exchanged by two separate streams of packets, one stream in each direction. The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.
In your previous posting you mentioned that you "set the minimum EXTernal rtp port at the sip tab". Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does. In my tests it didn't appear to affect the rtp port number used in a predictable manner.
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PSTN Line Tab:
NAT Mapping Enable: Yes
Sip Tab:
Handle VIA received: yes
Handle VIA rport: yes
Insert VIA received: yes
Insert VIA rport: yes
Substitute VIA Addr: yes
Send Resp To Src Port: yes
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http://www.voip-info.org/wiki/view/STUN
You are using CounterPath's XLite softphone. stun.counterpath.net is a STUN server on the list.
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Transferring two PSTN Lines from Office to Home

Hello
I am a bit confused since I am new to this stuff.
At my office I have two PSTN lines and I would like to transfer them to my home via a WiFi link that's already established. I am looking for the cheeapest reliable solution. From the reasearch that I have done so far I think I need 2 FXS ports on the House side and 2 X FXO ports on the Office side. I can install two SPA 2102 or SPA 3102 (newer model) on the Office side and plug the two PSTN lines that I have there. I should connect both devices to my WiFi router and configure them as two different subscribers with two different IPs from my subnet. On my other WiFi router at my home I should connect one LinkSys PAP2T which has two FXS ports and plug there two analog phone devices.
My questions are:
1. Does the above configuration sounds right?
2. Is there something that I should be aware of in the configuration of the devices?
3. Does the PAP2T support two different subscribers?
4. Will I have CLI on the analog lines at my home.
5. Where should I search for informations of such a configuration?
Thank you in advance
Solved!
Go to Solution.

"On the office side" the SPA2102 does not have an FXO port. The SPA2102 is essentially a PAP2T plus a router. You would need two SPA3102's for the FXO ports to interface the PSTN Lines.
"On the house side" you could interface it with a PAP2T which supports two phone lines (two analog phone attachments). You could also use the PAP2T and only use one of the phone lines if you only have one person answering the phones, i.e. forward the calls to a single address/number. The adapters support call waiting. You could also use either a two-line phone, or you could buy an ip phone such as a SPA941 or SPA942 and use a single or multiple ip lines.
The easiest way to configure it would be to use 3 (or 4) different voip accounts from a voip provider and send the calls from one account to another. As an alternative, it is possible to use direct ip addressing and not use a voip provider but the configuration is more complicated and the voip provider also has an advantage in that they can give you a perhaps more sophisticated voicemail than a simple home answering machine.
If by CLI you mean Caller Line Identification or Caller ID, the SPA3102 can pickup the incoming Caller ID and send it along, but if you are using a voip provider you will probably only receive the Caller ID of voip provider account doing the sending, although you might find an exception.
Some people find the configuration process frustrating, particularly the direct ip calling.

Trixbox SPA400 disconnect PSTN line after 60 seconds

Hi all, I just bought a SPA400 and am trying to connect it to my Trixbox. I followed the HOW TO guide in the voxilla forum, and in no time I was able to make outbound calls via the SPA400. This is my trunk set up
allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=192.169.0.71
insecure=very
nat=no
type=friend
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However, almost all of my calls got disconnect 60 seconds into the calls. I captured the packets, and it was the SPA400 that first sent the RTPC Goodbye message to my Trixbox.
Right after I power cycled the SPA400, things worked well for about 5 minutes or so.. then the hang ups. Does anyone seeing this problem? Can you please help me?
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I am having the same problem, very random though.
SPA400 disconnect PSTN line after 60 or more seconds Outbound Calls Only it seems.
I had the same problem with trixbox 1.2.
spa400 Firmware Version: 1.0.0.12
Netgear-FS750T2-Switch Managed with QoS Enabled between the Server and the SPA400
Phones Grandstream GXP2020 and GXP2000
OS Version: Linux asterisk.berkleigh.com 2.6.18-8.1.15.el5
#1 SMP Mon Oct 22 08:32:04 EDT 2007 i686 i686 i386 GNU/Linux
Asterisk Build: Asterisk 1.4.13 Asterisk GUI-version Revision: 1752
No indication of problem at asterisk console #asterisk -rvvvvv.
I am now monitoring the SPA400 Logs.
Things I see in Logs that I do not understand.
These are not in order and extracted from different parts of the logs. 2235862- SSMU 4 *** ERROR -- DOES not INCLUDE CNUM
3842- SSMU 4 SSMU: 3, ssmup_proc_connect attempted
40592- NMM 5 NMM: set_dccfg() calls tiuo_fxo_set_curlimit(1)
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Message Edited by berkleigh on 11-30-2007 12:37 PM
Message Edited by berkleigh on 11-30-2007 12:40 PM

PSTN lines to PBX via VoIP

Please be patience with me, I am relatively new in VoIP.
What I need is to replace an old RF telephone system by a VoIP technology using a wireless link.
There are 2 (two) PSTN lines that I think to connect to a 2 ports FXO Gateway.
This FXO GW will be connected to an Access Point 1 (AP-1).
The other Access Point named (AP-2) is 10 km from AP-1.
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The SPA3102's should work for you, but....
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Is SPA8800 working without PSTN lines when integrated with BE3000 ?

Hi Guys,
I have a SPA8800 configured with static ip address 192.168.1.2/24 and BE3000 with ip address 192.168.1.250 in the same network. I would like to connect a fax machine to SPA8800's fxs port however there are no PSTN lines connected to FXO ports , will this fax machine work?
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You can connect FAX machine to the FXS port of the SPA8800. There is no mandatory requirement to have FXO lines also connected to SPA8800. If you want to use E1-PRI line as PSTN connection for fax calls, that is fine as well.
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