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

Similar Messages

SPA 3102 & SPA2102 have too short dial tone time.

My 3102 has an "off hook" time of about 9 seconds before it times out if no key is pressed. This is usually OK, but my 2102 only gives me 3 seconds, which is definitely not enough.
I found in the Linksys "ATA Administration Guide v3 200809" :- ftp://ftp.linksys.com/downloads/NA/other/ATA_AG_v3_200809_AdminGuide.pdf
Syntax for the Dial Plan Timer
SYNTAX:
(Ps<:n> | dial plan )
s: The number of seconds; if no number is entered after P, the default timer of 5
seconds applies.
I have added this Ps command to my dial plans but while I can reduce the timer period, I can not increase them.
Both ATA's have the latest firmware installed and work as required in other ways.
Is their something else to be adjusted?

The regional tab:-
Call Progress Tones
Dial Tone:
Second Dial Tone:
Outside Dial Tone:
Prompt Tone:
Busy Tone:
Reorder Tone:
Off Hook Warning Tone:
All show ;10( as you suggested, but I've tried various longer and shorter times for both "Dial Tone" and "Reorder Tone" but no change from the 3 second timeout.I wondered whether the Reorder Delay which was set at 3, could be the problem, so changed to 10 with no effect.
Control Timer Values (sec)
Hook Flash Timer Min:
Hook Flash Timer Max:
Callee On Hook Delay:
Reorder Delay:
In case it is relevant, my dial plan is:-(13[1-9]xxxS0|1[38]00xxxxxxS0|190xx.!|0011xx.!|0[2-9]xxxxxxxxS0||<9:089>xxxxxxxS0 |<:089574>[2-8]xxxS0 )My software Version is 5.2.10It may be relevant that after the Dial Tone times out (in 3 sec) I do not get Reorder Tone but it just goes dead!RegardsMaurice

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.

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.

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 help

I 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

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.

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

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

SIP phone and SPA 3102

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.
Thanks

yytellmey 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.

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)

Spa 3102 : Setting up Voicemail

Hi Guys,
I have this SPA 3102 ATA and wondering how to setup the ATA configuration to use my phone voicemail instead of my ISP's voicemail.All incoming callers hear my ISP's voice message and are redirected to their message box. I want to use my Phone's message box instead. Thanks in advance.
-R

If you have a phone attached to the SPA with its own voicemail, there is usually a setting that you can make on the phone for the number of rings before the voicemail answers. This needs to be shorter than your voip provider's voicemail. Better yet, many (most?) voip providers have a way to disable voicemail at the account level and then you don't have to worry about their voicemail. This also solves the problem of a call going to the provider's voicemail when your phone is busy and you not realizing that there is a message there waiting for you.

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

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

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

Similar Messages

Maybe you are looking for