Spa-3102 Feature Question

Hello
I was wondering if the following setup is supported.
I would call a skype id and then using the spa3102, that id would use the fxo port and give a off hock status and then I would be connected to that line
is this supported/been done before?
thank you

I didn't read your questions carefully enough the first time.
The EXT IP: address is the external ip address of the SPA3102 itself, not the opposite one.

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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

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.

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 sending bad "From" field

I'm using some SPA-3102's (5.1.7 GW firmware) and am getting "Error 400: BAD REQUEST (multiple values in single-value header) FROM" responses from my PBX.
According to packets grabbed by wireshark, the INVITE from the 3102 goes out with the following FROM field:
From: MACKEY, RON E <sip:[email protected]>;tag=b9c8ff5d2616dc03o1
The ASCII version:
0000   46 72 6f 6d 3a 20 4d 41 43 4b 45 59 2c 20 52 4f From: MACKEY, RO
0010   4e 20 45 20 3c 73 69 70 3a 36 33 30 33 35 37 38 N E <sip:6303578
0020   31 33 35 40 31 39 32 2e 31 36 38 2e 32 2e 31 33 [email protected]
0030   32 3e 3b 74 61 67 3d 62 39 63 38 66 66 35 64 32 2>;tag=b9c8ff5d2
0040   36 31 36 64 63 30 33 6f 31 0d 0a                 616dc03o1..
My reading of the SIP RFC (and it's been a long time since I read an RFC) says that a display-name containing a comma must be enclosed in "".   This would seem to be confirmed by the bad request response from the IP-PBX.
Two questions:
1. Is this an error in the 3102?
2. In either case, is there any way to prevent the 3102 from sending the display-name?
Thanks,
Ron

Thanks for the reply.
Setting "Escape Display Name" to "Yes" had no effect.
In this scenario, the display name is picked up from the incoming phone company Caller ID, so I can't figure out how to blank it out without also losing the caller ID information.
I'm going from a POTS line --> PSTN 1 (on the 3102) --> 3CX (PBX software).   I'm using wireshark to verify that the packets coming from the SPA-3102 are just mirroring what's coming in from the phone company.   So, in my case, any name that comes in with the form "last name, first name" gets dropped by 3CX as improperly formatted.
Best regards,
Ron

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.

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.

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

Show more text on the featured question box

I made some featured questions for the Power BI site using Power BI Q&A but I can't get the box (small or large) to show the whole text. It is truncated before the where clause which is the important part to see when choosing which featured
question you would like to see! Is there a way to get it to display the whole text, or would I have to make my own way of displaying the options, then add the functionality by using the link instead?

I did find a way to put the "where clause" a little higher in the text.
Instead of:
Show community, total unique members, where community category is Technical
I phrased it this way:
Show Technical community category, community, total unique members
and the bold is what shows on the featured question tile on the main page.

Syslog not being sent by spa-3102

I can't seem to get any syslog to come out of my spa-3102. I've enabled syslog putting the ip address in both the syslog server and the debug server in the "System" tab. I set the debug level to 3. I have my system in a "double NAT" configuration: internet -- spa-3102 -- NAT router -- LAN (incl. syslogd) I don't know how the SPA chooses which interface to send its syslog UDP to. Since my syslog server is on my LAN, I assume that I have to specify my router's external IP (assigned by the SPA-3102), and then I configured my router to forward the packets to my LAN machine running syslogd. I reconfigured my linux syslogd to accept remote syslog and to log local0 and user. Kind of complex I know, and multiple points where it could be failing. But I want the SPA in front so it can do QOS and so I don't have to worry about NAT for its SIP support. Any suggestions on what I might do.

Just responding to myself -- it seems it was some problem with my particular syslog server, nothing to do with the SPA. I set up an alternative network syslog on a different LAN machine, and it is working fine through the double NAT and forwarding of UDP 514. -mda

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

Configuring SPA 3102 with Ooma Telo Problems

Does anyone have any experience with configuring the SPA 3102 with the Ooma Telo devise. I have been around and around on this with no luck. I have been able to get into my 3102 ip address fine, but I need a proxy server to configure the 3102 with my existing Ooma device. Ooma doesn't seem to have one to dish out. Ooma is also saying that they don't have a portal for configuring external devices to the Ooma.
Has anyone ever tried to connect the two? I'm just trying to cut my losses here and move on if this is futile.
Thanks anyone!
Duncan

Wrong forum, post in "small business voice - SPA phones". You can move your post using the actions panel on the right.

SPA-3102s back-to-back

Hello, we are trying to use two spa-3102 boxes in a back to back configuration to provide phone lines to a construction site (via a Cisco wireless bridge) like this:
(PSTN)--(SPA3102 FXO PORT)--(Wireless Bridge)--(SPA3102 FXS PORT)--(Phone/Fax)
We have voice calls passing correctly in both directions, and the phone/fax machine at the construction site is able to receive faxes most of the time, but they are unable to send outgoing faxes. When the fax machine attempts to dial, you can hear the machine pick up, dial the number, and then there is a few moments of silence followed by a fast busy signal. Any ideas?
Thanks!

EvilFetus wrote:
When the fax machine attempts to dial, you can hear the machine pick up, dial the number, and then there is a few moments of silence followed by a fast busy signal. Any ideas?
Thanks!
If the dialing waits until the interdigit timeout and then gives a fast busy it usually means that the digits dialed are not correct. If the dialing is "back-to-back" as you say, you probably are dialing some code that sends a sip uri for direct ip dialing. There are a number of variables involved.
You should be able to view the last dialed number on the INFO tab.

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 Feature Question

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