HELP: SPA-3102 Gateway Setup Question

Hello,
I would like to set up the SPA-3102 to do the following:
1. The adaptor is registered to one SIP account
2. The adaptor can make outgoing calls through two or more SIP accounts which are not registered.
3. Calls to local numbers and emergency numbers are routed to PSTN
4. Calls to SIP phones are routed to the registered SIP account
5. Calls to long distance and international numbers are routed to the registered SIP account.
6. If we dial with a prefix, calls to long distance and international numbers are routed to the 2nd SIP account which is not registered.
Currently, we have 1, 3, 4, and 5 working. But 6 is not working. Is 6 possible? If so, could someone help me with an instruction of how to set it up?
Thanks,
AVS

You setup the second sip account in one of the gateway fields. Let us assume you are using gateway 1, the sip provider is voipbuster, and your userid is avs. Gateway 1:
[email protected]
GW1 Auth ID: avs
GW1 Password: your_password
GW1 NAT Mapping Enable: (same as you have on Line tab)
In the example above the provider's sip proxy is sip.voipbuster.com. In the example above avs and your_password are the userid and password for a specific account at the provider.
You put a prefix element in the dial plan, let us assume you put #8 for the prefix and wish a 2d dial tone after you dial #8:
|<#8,xx.<:@gw1>|
You dial #8 and you get a new dial tone, you dial the number and the number is sent to the gateway provider for call termination. The sip provider that you use must allow you to make calls without registration. Some providers don't allow making calls without registration.
Message Edited by hw on 06-12-2008 11:45 AM

Similar Messages

10.8 Server (VPN Secure Internet Gateway) setup question

I am running Mountain Lion 10.8.4 with Server 2.2.1
I am attempting to setup the server to allow connection to my internal/Private LAN
I have the source (External Internet access) setup as #1 in the service order (en0)
and the Private network as the secondary (en4)
I followed the steps on http://macminicolo.net/mountainlionvpn and input my own IP's when needed
I am able to connect and authenticate to the vpn and able to get internet access through the vpn
unfortunatly I am unable to reach anything on my private LAN
this is my settings in my customNATRules:
nat on en4 from 10.0.0.0/24 to any -> (en4)
pass from {lo0, 10.0.0.0/24} to any keep state
i have the sysctl.conf setup with
net.inet.ip.forwarding=1
I also changed the com.apple in pf.anchors to reflect the instructions above
Network Settings
(en0) My external ip is 192.168.168.4 to my firewall (not giving you my full outside)
and the DNS Server is pointing to itelf via 127.0.0.1
(en4) My Private LAN is set with the DNS to my private DNS servers
VLAN is setup with the same settings as the instructions state in the link above and I have the DNS set as 127.0.0.1
DNS Server Settings
I have my DNS server configured with my local hostname with the Vlan, internal ip, and external ip pointing back to the hostname.
i have the forwarding DNS servers configured to my private DNS servers for the private lan and as the 3rd I have 8.8.8.8 for general internet
VPN Server settings
I have the host name and shared secret set
I have 10 IP's for client addresses with the same IP segment as the VLAN
DNS settings i have routed back to the gateway of the vlan
I have one route configured i am using in my private lan to be routed private
is there anything I am missing or setting up incorrectly? I am struggling at this point and need some help.
if you need any more info please let me know

The instructions on that web page aren't applicable to your case. Don't follow them.

200 OK message before call is established with linksys SPA 3102

I recently bought a cisco linksys SPA 3102 gateway to help me forward incoming VOIP calls to the PSTN network via the PSTN line. I also installed syslog to catch the sip trace. When i placed a call, after the SIP Invite and Trying, I immediately get a 200 OK reply from the PSTN LINE, just as soon as the calls is forwarded to the PSTN network for dialing. This 200 OK reply triggers the billing from the SIP side mean while the call has not yet been established.
Is there a way to stop this per-matured 200 OK reply from happening?
I will be very grateful for your help or hints.
Cheers
Emmanuel

I recently bought a cisco linksys SPA 3102 gateway to help me forward incoming VOIP calls to the PSTN network via the PSTN line. I also installed syslog to catch the sip trace. When i placed a call, after the SIP Invite and Trying, I immediately get a 200 OK reply from the PSTN LINE, just as soon as the calls is forwarded to the PSTN network for dialing. This 200 OK reply triggers the billing from the SIP side mean while the call has not yet been established.
Is there a way to stop this per-matured 200 OK reply from happening?
I will be very grateful for your help or hints.
Cheers
Emmanuel

WRTU54G-TM/SPA-3102/Asterisk Disconnect Tone/Busy-Reorder tone?

I have a setup where I'm using the T-Mobile@Home Router (WRTU54G-TM) as a Trunk on my Asterisk system (PIAF). The WRTU54G (Phone 1 Port) is connected to the FXO (Line) port of the SPA-3102. I can making outgoing calls without any problems. However, incoming calls to my T-Mobile@home number once it hits the voicemail system on the Asterisk system and if the call hangs up before or after leaving a message, the "system" does not release the line and not do so unless I physically unplug the phone cord from either port (SPA-3102 or WRTU54G-TM). If I answer the cincoming calls and either party terminate the call, there is no disconnect issues; only when the call goes to voicemail. Is there any changes I can make to either the SPA-3102 or Asterisk, that will solve this problem/issue?
The problem seem to be related to:
a) CPC isssue and/or
b) Busy/reorder tone and/or
C) Disconnect Tones (does anyone know what the specs are for the T-Mobile system? Looks like this: 480@-30,620 @-30;4(.25/.25/1+2))
I saw on another site where an individual was able to do this:
..."Im running FreePBX on Asterisk and was able to use the busy/reorder tone by editing some lines in my zap channel config files. My solution was to simply program the PBX to detect that busy tone that T-mobile's @Home router makes after the call has ended, and use that as a signal to know when to hang up. Worked excellently, although the tail end of our voice mail message usually records a couple seconds of the busy signal... which I decided was not worth worrying about."..........
Not sure how I would implement a similar scheme, since I'm not using any ZAP channels or digium cards. Any help or suggestions welcome!

You could try to adjust this options on your SPA3102 PSTN Line. Under PSTN Disconnect Detection.
PSTN Long Silence Duration
This is minimum length of PSTN silence (or inactivity) in seconds to trigger a gateway call disconnection if <Detect Long Silence> is yes.
The default is 30.
Try to lower the values.
And Also PSTN Silence Threshold:
This parameter adjusts the sensitivity of PSTN silence detection. Choose from {very low, low, medium, high, very high}. The higher the setting, the easier to detect silence and hence easier to trigger a disconnection.
The default is medium.
Regarding for the 480@-30,620 @-30;4(.25/.25/1+2. basically this it the default settings for the US Disconnect tones. No need for you adjust.
Hope this help

Linksys SPA 3102 not detecting hang up from Asterisk FreePBX

Hi,
I am forwarding calls from my pstn to freepbx using SPA 3102 (pstn to voip gateway). I have programmed asterisk to disconnect the call. It seems asterisk is working fine and disconnecting but pstn user still hear the call ringing. How can i make the call disconnect?
Thanks,
Rajeev

It's not be possible to solve issue described by rajeevraj22 in first message. According description, the incoming call has not picked up ("pstn user still hear the call ringing"). Unfortunately, POTS signaling protocol doesn't allow to reject the incoming call. SPA3102 can either pick up call or ignore ring signal on POTS line. In the second case the calling user is hearing ring back tone until he hang up or call setup timeout occur.
Despite your's (ildefonso_v1) problem seems not to be same (your call has been picked up first) there may not be solution available as well. CPC stands for "Calling party Control". CPC is signal sent from terminating PBX to called phone to indicate that the calling party has hung up. Terminating device (SPA3102 here) is recipient of CPC signal, not the source. E.g. CPC is related to opposite signaling direction that the one you are speaking about.
The call disconnect from end device to PBX is signaled by high impedance of end device.
I see when spa line is Idle the current Voltage is 52V, When a call is done the voltage is -7V. When I finish a incoming call from any internal extension or from asterisk the voltage is 52V again, but the call does not hang up in the other side.
OK. So idle line voltage is 52V. According your description, the SPA3102 disconnect properly from line on end of call (voltage rise to 52V). If the call doesn't disconnect, then it's not matter of SPA3102 but matter of terminating PBX.
The behavior like it is not so common in current phone networks, but older phone network switches allow no hangup from called side at all. Only caller is allowed to terminate the call.
All at all, you can't solve the issue from your side of wire. If terminating PBX is not willing to disconnect call immediately, you can do nothing with it. Ask your Telco operator for support. The behavior of particular line may be configurable. But don't put so much hope on it.
After 30 seconds I can see in the syslog POL REV -47 52 then the call hangs up in the other side.
It is the CPC signal from PBX to you. The PBX considered the call is over and is signaling it to you.
Rate helpful responses. It will help others to found solutions.

SRTP Behind firewall (SPA-3102)

Hello,
I am using an SPA-3102 in the following setup.
Internet->FTTH Converter->Apple Airport connetcted through PPPOE->SPA-3102
In this setup, I am unable to make SRTP calls. I.e. Even if I dial *18 + number, to another device with certificate, I cannot hear the three unique tones to affirm that the call is encrypted. Although, I am able to receive SRTP calls.
But when I did this: Internet->FTTH Converter->SPA-3102 conntected through PPPOE
It is possible to initiate SRTP calls, i.e. I can hear the three unique tones when I dial *18 + number.
My questions are:
Is it necessary to unblock some ports to SRTP to work?
Tell me also about other possible factors that may block me connecting thus.
Regards and thanks in advance.
Message Edited by sixties on 01-03-2008 07:18 PM

Try to set the NAT mapping option to yes on the LIne tab. This may help in the firewall traversal.
also on the SIP tab there is an option for Send RESP to SRC port. set this to yes.

Connect back to back 2 SPA-3102

Hello everyone,
I would ask a little help from any expert in order to fix the following connection:
(analog phone)--(SPA-3102)--(Access Point)--> <--(Access Point)--(SPA-3102)--(analog phone).
First of all is it possible to do it ? And also do I need any special setup in the SPA-3102 ?
The story is to transfer an analog line through air in another place near my home which doesn't have any analog lines at all.
Thank you in advance
vangellis

There are probably several ways you can configure it. I will outline one way that will ring the distant spa on an incoming pstn call and will return a pstn dial tone to the distant spa when the attached handset goes off hook.
The distant ata is a SPA3102. As an alternate you could also use a different but similiar Linksys adapter such as a PAP2.
On the SPA3102 attached to the PSTN line.
PSTN Line Tab
Line Enable: YES
NAT Mapping Enable: yes or no depending on the router used
NAT Keep Alive Enable: yes or no depending on the router used
Register: No
Make Call Without Reg: Yes
Ans Call Without Reg: Yes
UserID: spaone
VoIP-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: none
VoIP Default DP: 2
VoIP Caller Default DP: none
none will give you the actual PSTN dial tone when the VoIP-to-PSTN gateway answers the call
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no
PSTN Caller Default DP: 2
Dial Plan 2: (S0<:spatwo@ipaddress: 5060> )
The dial plan will automatically dial the distant SPA3102 Line 1 when the PSTN-to-VoIP gateway answers the call
Where spatwo is the userid on the Line1 tab of the distant SPA3102 (spatwo)
Where ipaddress is the ip address of the distant SPA3102
Where 5060 is the sip port number of the Line 1 tab on the distant SPA3102
VoIP Answer Delay: 0
PSTN Answer Delay: 2
Where 2 seconds is long enough to receive an incoming caller id on the pstn line (if you care about it)
Line-In-Use Voltage: 30
Where 30 is about half way between the on-hook and off-hook voltage of your pstn line (polarity disregarded)
Line 1 Tab
Enable IP Dialing: YES
On the Distant SPA3102 Line 1 tab
Line Enable: Yes
NAT Mapping Enable: yes or no depending on your router
NAT Keep Alive Enable: yes or no depending on your router
Register: No
Make Call Without Reg: Yes
Ans Call Without Reg: Yes
UserID: spatwo
DialPlan: (S0<:spaone@ipaddress:5061> )
Where spaone is the useid on the pstn line tab of the SPA3102 attached to the pstn line
Where ipaddress is the ipaddress of the SPA3102 attached to the pstn line
Where 5061 is the sip port number on the pstn line tab on the SPA3102 attached to the pstn line
Enable IP Dialing: YES
Note: Depending on your network you may or may not have to have NAT Mapping Enable. If you do have NAT Mapping Enable you may also need to configure a STUN server depending on the router that you are using.
If the two ip addresses are not on the same local network you will need to forward the sip port in the router to the SPA3102 and you will need to use the external ip address.

SPA-3102 and Fax

Here's the problem: I am currently using a fax-switch that answers the incoming line, listens for a fax tone and, should it hear it, forwards the call to a fax machine. Without fax tone, the call is routed to the SPA-3102 and treated as voice.
This setup works nicely, but has one BIG disadvantage: All fax switches 'steal' the Caller ID. I am now trying to skip the fax-switch and use the SPA-3102 directly, by connecting the fax machine directly to the phone port of the unit. Since the SPA-3102 has the ability to recognize incoming faxes, is it able to route the call directly to the phone port? Without actually bothering the connected VOIP equipment?
I have tried to find a solution all over the Internet, but I seem to either be to blind to find anything, or, it might just not work. Thanks for your answers and suggestions.
Michaela

Thank you. I knew there must be a quick fix. Though ring thru would make the fax machine take all calls, which would make incoming phone calls be lost. If things were that easy, I wouldn't have bothered to ask. I was expecting somebody with actual Linksys knowledge to answer my question. Thanks again.

Connecting an SPA 3102 after the computer

I am moving to a house in a fairly remote part of Australia without a telephone connection. To get the phone connected will be very expensive and take considerable time. I have mobile phone and wireless internet connection available but only with the most expensive provider. I am not a big user of the phone but every now and then, I have to make one of those calls that go through adozen menus and then leave you on hold for an hour, theb type of call that can send you bankrupt using a mobile service.
My proposed solution is to have a prepaid wireless internet account, using a USB dongle connected to my computer as the internet connection. Then to connect either a SPA3102 or a PAP 2T between the computer and a handset and use a VOIP service such as Voipcheap to make calls. With this setup, I would have unlimited landline calls within Australia for around $50 per year, much less that something like Skype.
Can I do this and if so, how? I have both devices and they are both presently configured to work connected to my Linksys gateway. Do I connect the computer to the LAN or eithernet of the SPA 3102 and will it just require a normal LAN cable? How do I assign the various addresses required? Do I need any extra software for my computer?
John Adams

Hi John -- Thanks for participating in the Small Business Support Community. Please consider posting in the section dedicated to Australia/New Zealand here:
https://supportforums.cisco.com/community/netpro/small-business/international/australia_newzealand?view=discussions.
Thanks,
Stephanie Reaves
Cisco Small Business

Create line extension between two SPA-3102

I`m having problems to create a line extension between two SPA-3102
I have one SPA-3102 connected to an analog PBX system with IP 192.168.0.201, and the other SPA-3102 with analog phone and IP 192.168.0.200
I succesfully setup them to make a call from the first to the second
But I couldn`t setup them to make a call from the second (192.168.0.200) and give me the dialtone of the PBX connected to the first SPA-3102 (192.168.0.201).
I could setup a hot line on the second SPA-3102 (192.168.0.200) and call to 192.168.0.201, but it doesn`t take the line to hear the pstn dialtone.
I saw many answers about this problem, but no one resolve the problem, i have the latest firmware. please, anyone could help me and if it`s possible to work please send me all the configuration needed.
Thanks again

Hi Jeremy,
I have a similar problem, I have one PSTN line (say Line1) with free minutes to mobiles, so its good for outgoing calls. The other line (say Line2) which i have is acually VoIP but it comes with its own hardware (magicJack if you have heard) so I can't use a SIP client and have to use the supplied Hw client, but it does give me an option to connect any normal phone to this magicJack (i suppose that would make it a fxs port). Now this magicJack is cheap for other people to call me.
I want to find a solution so that all the calls I receive on Line2 get forwarded to my mobile number via Line1. And if I receive any calls on Line1 they should be treated normally (my home phone rings). Do you have some idea how I can achieve this with minimal spend? Thanx
Atif

SPA 3102: loosing connection intermittently

Hi,
I am using SPA 3102 and D-Link wireless router together. Connection is Cable Modem to SPA 3102 to D-Link DIR-615.
Both the phone and wirless works fine in this set up. The main issue is when I use utorrent, SPA 3102 keeps resetting (loosing connection intermittently). All the three lights goes of for a second and then comes back. End result is, I have to keep resetting the routers (power off for 10 secs and switch on). This is really annoying, I have to keep doing it atleast 3 or 4 times a day. I guess the problem might be with the heavy downloading/uploading with utorrent. Any way out ? Help appreciated.
aj

Ooops, sorry My first post was a mistake. Here is my question:
I have modem connection to a remote site.This connection must go from wire to Ethernet. The modems are 2-wire, not dial-up, and are embedded in a device, so there is no access to the serial port of the device.
I have two SPA3102. On the remote site the modem is always connected to the FXS port. This port is configured with SAS Enable=Yes and SAS Inbound RTP Sink = $IP. On the local site when modem is connected the conection is automatically established. With Wireshark I can see that RTP audio stream is flowing in both directions but signal can not be heared on the server side. What should I do to hear the inbound stream? Is it possible with SPA3102?
First I tried with SPA112 as SAS on remote site. The signal passes in both directions and can be heared on SPA112 port. Same settings. But SPA112 causes delay of 140ms and timing is critical for my devices. SPA3102 has only 80ms delay with RTP Packet Size = 0.02 and modems work with two SPA3102 configured as hot line. The problem with hot line is that in case of connection fail I have to go to the remote site to disconnect the modem and connect it again. Not very convinient.
Thanks
Svilen

SPA 3102 Admin Guide

Where can I download the Admin-Guide for the Linksys SPA 3102 ?
I don't remember how many times I tried to find such a link and how many hours I spent so far with it. I could find answers to this question but the provided links were not valid anymore.
Thank you for your help.
Solved!
Go to Solution.

Finally I managed to find a working link in this forum. Why can this document not be made easier avalable? It would have been an advantage if it could have beeen found by using the search on the Cisco home-page.

SPA-3102 Caller number not seen

Hi ,
french user of a SPA3102 , i use this gateway between my phone port on DSL Box and a 3cx extension. It works for all the calls incoming or outgoing.
The only problem i have is about caller number identity, what is the parameter i have to modify to have transmission in INVITE field in 3CX and know which number calls me.
For now i just obtain my default 3cx number extension or my phone line number.
I hope to be clear because my english is not enough good to explain with lots of technical details.
Waiting for ideas, Thanks in advance,
Best Regards
Profwalken

I only wish that were the case in the UK. I cannot get Caller ID working. That last PSTN caller is not shown on the spa3102 web/admin/voice/advanced page internally the last pstn called number is shown.
The last previous firmware was 3.3.6 currently the spa3102 has
Product Information
Product Name:
SPA-3102
Serial Number:
FMxxxxxxxxx
Software Version:
5.2.13(GW002)
Hardware Version:
1.4.5(a)
MAC Address:
xxxxxxxxxxxxx
Client Certificate:
Installed
Customization:
Open
I can post any config needed if someone can assist me with this. I have read about everything I can find on the web over the last few weeks and cannot find an answer to this.
I MUST STRESS THAT I AM IN THE UK so US or other solutions do not work!
Any help or assistance appreciated.
Best regards
Anthony

Receiving fax trought spa 3102 asterisk 1.4 and spa2102

I have some problem receiving fax from a pstn line trought spa3102 - asterisk and spa 2102. I have updated the firmware of the latest version for spa3102 and 2102 and I have setting the port of spa 2102 were is attached the fax machine as described in the CISCO Small Business Administration guide (page nr. 55 - using a fax machine) but i cannot receive fax at speed upper to 2400 b.
My setup is:
pstn line -> spa3102 - asterisk 1.4.29 -> spa2102.
The spa3102 is configured to pass all control to asterisk. When a call is answered and it is a fax, this call is transferd by operator to the internall extension of spa2102 where is attached the fax machine.
I don't know as configure the pstn port of spa3102 for fax receiving.
Can you help me ?
thank in adavnce.
e.

Thank you for your reply.
Ok, I report my solution because I think that can be userfull for other users.
I have resolved the problem in this manner:
1) on the SPA 3102 I have set the PSTN Line, audio section:
a) preferred codec G711u;
b) use pref. codec: YES;
c) silence supp. enable: NO;
d) echo canc enable: YES;
e) echo canc. adapt enable: YES;
f) echo sup enable: NO;
g) FAX CED Detect Enable: YES;
h) FAX CNG Detect Enable: YES;
i) FAX Passthru Codec: G711u;
j) FAX Codec Symmetric: YES;
k) FAX Passthru Method: reinvite;
l) FAX Process NSE: YES;
m) FAX Disable ECAN: no
2) on the SPA 2102 on the line were is attached the fax machine, audio section:
I have followed the setup as described in the document that I have mentioned in my first message that is:
a) Preferred Codec: G711u;
b) Use Pref Codec Only: YES
c) Silence Supp Enable: NO
d) Echo Canc Enable: NO
e) Echo Canc Adapt Enable: NO
f) Echo Supp Enable: NO
g) FAX CED Detect Enable:YES
h) FAX CNG Detect Enable: YES
i) FAX Passthru Codec: G711u
j) FAX Codec Symmetric: YES
k) FAX Passthru Method: reinvite
l) FAX Process NSE: YES
m) FAX Disable ECAN: NO
n) FAX Enable T38: NO
o) FAX Tone Detect Mode: caller or callee
With this configuration I can receive and send fax to 14.400 on PSTN line acros the asterisk and the SPA 3102 / 2102 device.
The phone attached on the line 1 port of the spa 3102 have not echo when answer or call on the PSTN line as the other phone that have access to PSTN Line.
The phone of the fax macchine have echo, but usualy is not used as phone.
E.

Remote Provisioning for SPA 3102 and PAP 2 Series

Hi,
can some one help me with SETTING up Provisioning server for SPA 3102 pap2 etc using TFTP servers (I want a detailed explanation ) please it is urgent

These urls might help:
HTTPS Based Remote Provisioning with the SPA2102, SPA3102, and SPA9000:
http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a33b6a.shtml
http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa3102/release/notes/SPA3102_RN_V5-2.pdf

HELP: SPA-3102 Gateway Setup Question

Similar Messages

Maybe you are looking for