Spa 3102
Salve,
ho da poco acquistato l'apparecchiatura di cui in oggetto, non essendo pratica di queste cose, non riesco nemmeno a fare la prima fase, cioè digitando l'IP 192.168.0.1 non "parlo" con il Gateway, il mio pc sembra impostato correttamente sul rilevamento automatico dell'IP.
Stò sbagliando io o debbo impostare o fare altro ??
Ripeto che non sono un esperta informatica.
Grazie per L'attenzione
Karin
These forums are mainly english speakers, and the SPAs are covered under the SMB forum.
Similar Messages
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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 -
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.
MichaelaThank 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.
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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
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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 urgentThese 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!
DuncanWrong forum, post in "small business voice - SPA phones". You can move your post using the actions panel on the right.
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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 AdamsHi 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 againHi 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 Line is disconnecting every 10 minutes and 40 sec
Hi,
I've a SPA-3102 connected behind a Motorola router, for any reason I'm not able to find, each time you call someone, after 10 min and 40 sec, the line cuts, and you have to call back.
Software Version: 3.3.6(GW) Hardware Version: 1.4.5(a)
SPA-3102 has a fix IP, and config has been done by my phone IP provider, does anyone have an idea which settings needs to be modified to avoid this cut?
Thanks for your helpI think under the “Regional” tab, you will have “Control Timer Values” and I belive thios is where you have to adjest the timer. I am not quite sure as well what parameters are needed to tweek in this section but I am sure that it should be one of them. Maybe try adjusting the “Reorder delay”. I suggest contacting Cisco Tech support to further look into your concern. I believe this unit belongs to the business series devices that Cisco is now supporting. Try to go to this link for the other business series devices and the site where you can get hold of Cisco for support:
http://www.cisco.com/web/products/linksys/index.html -
Disable calling name presentation on SPA-3102
Hi,
If I send a SIP INVITE to my SPA-3102, where the From header is like this -- (spaces inserted to stop the forum software treating it as an email address -- they're not there in the real invite)
From: Caller Name <01234567890 @ my.sip.server.net>;tag=as4b617ab1
-- the SPA-3102 generates a Caller ID spill on its FXS port with 'Caller Name' as the calling name, and '01234567890' as the calling number. That's all well and good.
If the From: header doesn't have a caller name, but is like this instead --
From: <01234567890 @ my.sip.server.net>;tag=as4b617ab1
-- the box sets the calling name to be 01234567890 as well.
Is there any way to turn that off, and have the SPA just not present a calling name at all?
If not, no bother! I'm just trying to get my box to behave a little more like BT with regards to caller ID presentation -- they don't ever send a reason for no calling *name*, but if the calling number is withheld or unavailable they will set the calling name to Withheld or Unavailable -- and set a reason for no calling number.
Many thanks!
Martin
Message was edited by: Martin Thorpe -- hopefully removed the auto-'email address' tagging! (Argh, no, it didn't. Bodged a different way.)Hi Lindsey,
Thanks for the quick response. Here's a complete SIP invite -- I've changed the telephone number and put spaces around @ signs again, but everything else is unmodified.
INVITE sip:spa-line1 @ 81.2.113.115:5060 SIP/2.0
Via: SIP/2.0/UDP 81.187.239.177:5060;branch=z9hG4bK4062e0e9;rport
Max-Forwards: 70
From: ;tag=as75e22314
To:
Contact:
Call-ID: 445f75c33908fff74829a514159e9946 @ sentry.met24.net
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Mon, 29 Oct 2012 19:51:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286
So there is a contact field in there as well.
That's from a slightly patched Asterisk server, which doesn't put a calling name in if it's blank -- by default if you didn't set a calling name, Asterisk will also set the calling name from the calling number and you'd get this instead:
From: "01234567890" ;tag=as54c7bb08
I've done product management myself so I know one customer asking for it to work a little differently (as opposed to it doing something wrong!) isn't going to make a change -- that's no problem at all. If it were to be changed, I'd rather the ATA didn't generate a calling name field in the CLID spill at all, rather than 'Unknown'. But hey, that's just my opinion!
For the avoidance of doubt, the ATA is always generating the calling *number* field in the CLID spill correctly.
Thanks again!
All the best,
Martin -
Hi,
I don't have an ipPBX or call signaling server. Can I register a SIP phone on the remote SPA 3102 then call the remote number. SPA 3102 is on remote site, the FXO port is connected to a phone line. My SIP phone is on local site, connected to Internet.
Thanksyytellmey wrote:
Hi,
Can I register a SIP phone on the remote SPA 3102 then call the remote number. SPA 3102 is on remote site, the FXO port is connected to a phone line. My SIP phone is on local site, connected to Internet.
Yes you can do that. You need to know the ip address where you are calling. This is called direct ip dialing. You can call the SPA3102 and have the attached phone ring, or you can call the SPA3102 and have it dial a call out the pstn line. It all depends on what you want to do.
Initially you can get it working with whatever ip address you have at the moment. For the long term, if you don't have a static ip address you can get a symbolic address from someone like dyndns.com and then when your ip address changes you setup some means, either thru your router that supports dynamic dns, or with a pc program to keep your ip address updated at dyndns for your symbolic address.
You almost always have to forward the sip signalling port in your router to the SPA3102. You may also have to forward the spa's rtp ports in your router to the SPA3102.
There are a couple of ways to configure the SPA3102 when you want to bridge the call out the pstn line. The simple way is to just return a dial tone to the caller and then the caller enters the pstn number they want to dial. A more complicated way is to send a sip invite to the SPA3102 and have the SPA dial the number. The latter method is more reliable because you don't have to send dtmf signals over your voip link. -
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
EmmanuelI 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 -
SPA-3102 - phone port dead (FXS?)
I have a SPA-3102 that one day stopped providing dial tone to the connected telephone. I can still see it on the network, configure it, and all looks fine there, but the connected phone gets no dialtone, no voltage, touch tones don't work, no IVR, etc.... I've tried several phones, so I assume it's the adapter. I've tried to find out how to get it RMA'd and sent to Linksys for repair, but every avenue I've tried to pursue tells me that Linksys doesn't support it and i have to go through my reseller... The reseller says to go the manufacturer. Can someone please tell me: A) Am I missing something simply on this problem? B) WHO to contact at Linksys to get a replacement. Thanks! Steve
for one, calls made to the PSTN line form the SIP or internet will definitely not ring the FXS port. You must have the 2nd account dedicated to the call going to the PSTN only,
these are the call flows for the SPA3102
incoming VOIP
VOIP to FXS
VOIP to FXS hop over to PSTN
VOIP( THRU PSTN line) to FXO( PSTN)
Out going VOIP
FXS to VOIP
FXS to GW0
PSTN ( FXO) to VOIP
PSTN to FXS ( PSTN RINGTHRU LINE 1) -
I am seriously stumped.....I have placed my SPA 3102 between my cable modem and my apple router. I would like to forward some ports from my spa 3102 to my apple router. I've gone into the admin console and defined the settings I want on the "Application" tab. Yet there seems to be no difference when I submit all changes. The darn SPA just doesn't seem to actually forward anything. After several hours banging my head against the wall I'd greatly appreciate any assistance. Thanks, -Chris
I agree the ideal solution is one where the Sipura is behind the Apple router. In fact I had previously been in the setup with my previous Netgear router. Currently there is a bug in the Apple Time Capsule that prevents the defining of a range of ports (ie. 10000-20000) therefore I was originally unable to get the setup to work. I have managed to get the Sipura to work behind the Apple router by enabling NTP-PMP on the Apple router. I'm a little concerned about this setup as I'm not familiar with how secure this setup is. I had at one point tried to set the Apple router in the DMZ on the Sipura and it always amounted to nothing. The darn Sipura just wouldn't ever allow for ports to be forwarded. In any case, thanks for the input.
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Hi I have the SPA 3102
How do I backup the settings and then restore them?????
ThanksYou can backup the SPA3102 settings by logging into the SPA3102 as Admin/Advanced and then saving the web page to your hard disk drive using your web browser (MSIE, FireFox, etc). A single save will save all the tabs under the Voice tab. Password fields are not saved. You have to do a separate save for the Router tabs.
There is a technique to modify the saved web page which will allow you to restore the web page (without the passwords). I haven't used that for awhile so I am not sure about the details.
Message Edited by hw on 10-28-2008 06:39 AM
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