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

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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.

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

Strange dial tone

Hello, just noticed recently when I pick up the phone I have a strange dial tone.
It sounds like the normal one but then the tone lowers, then comes back up, then down again. Kind of like a siren on a police car/ambulance only not as loud.
Any idea what could cause it, there is only one socket in the house, and we only bought a new phone back in October, broadband speeds all seem fine and I have removed the sky box to also see but its still there.
We have an overhead line and some quite heavy rain a few nights back, was wondering had the connection box on the side of the house got damp and would I be charged for someone to come and look at it?
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Hi,
I've been researching this issue on the internet and came across this post as it sounds exactly what has happened to us.
On bank holiday monday, we set our phone on divert #21# as usual when we spend a day out. We came home only to find we had a strange ring tone (sounds like a siren) and we can't make any outgoing call, nor can we switch off the divert. Even though we can't make any outgoing calls, the divert still works so we can take calls on our mobile.
It is very strange. We have reported the fault with BT who have checked the line remotely and say it looks fine. We have test our internal equipment, and also tested a phone through the emergency socket which runs into the master socket. We still get the same ring tone.
BT.com/faults suggest now that it may be something not right near our home.
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This topic has been discussed plenty of times before, please search CSC for answers before posting.
The cause for this is overlapping in your dial plan, unless every single option after the first digit has the provide secondary dial tone (and that means, every single DN or pattern in your dial plan), you won't get it there. Look at overlapping in your dial plan and fix that overlapping to get it at the right time.

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:
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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

I have a wireless home phone connect device. Can call out on it when dial tone present, but most times cannot receive calls since purchase. Anyone else having a problem with their home connect device?

Have had Wireless Home Phone connect device refreshed and updated three times since I purchased it in September 2014. Last tech I worked with on it initialized an NRB to check the network. I checked the location of the tower, which is less than one mile up the hill with clear line of site. The device always has two antennas on it. The device is great when it does work, but most of the time it rings three times and then jumps to an automated message saying the person I am calling is not available. I've actually gone up there several times to try an fix the device, but always end up calling a Verizon tech who has me unplug it, pull the battery, and they do some network stuff and then we turn it back on and it works. After the last time on 10/02/14, I called it and got the message. I drove up there, 100 miles one way, and called it while standing in the room. Got the same message. I picked up the phone, heard a dial tone and dialed my phone and it range through. I hung that phone connected to the device and call it using my cell phone. It rang through. I went home and tried to call it the next day. Again it did not ring through and I got the message after three rings. I call another phone at the facility and had them call me on my mom's phone and it is fixed it again. The fix last for a day or two and then it's broken again, like it is today. If it's a network problem then it must be fairly close to the area because no matter what we do it fails again. So much for the 99.98% reliable. This thing is a piece of junk!
I purchased this device and put it on my plan so that my mother who we placed in a home at 96 years old would have a phone connection that she could use and wouldn't flip her out. My family and her family has not been able to faithfully connect with her at this critical time in her life. I am giving Verizon 10 more days to figure this out and then I am going to ask for a total refund not to mention how much time and travel I have put into this piece of junk. Either the home connect transceiver is broken or the network is broken. Why should I pay close to $70 for a service that has not worked more than several days out of 35 days? Also I have spent close to three hours on the phone with decent technical people, but it's my time. I should send Verizon a bill for every time I have gone up there to fix this device as well.
What perplexes me is that a regular cell phone would have sufficed, but because Verizon has to have a data connection that is not available on a regular phone and it was going to cost me $30/month, I decided to go with the home connect. Seemed like a nice solution for an older person who would have a hard time using a cell phone and keeping it charged. Plus after all the fees for getting it connected, it only cost $20/month. That would be a pretty good thing, IF IT WORKED....... The other thing that irks me is, why do I have to pay for a data connection when the cell phone Verizon originally provided can't even access the network. What ever happened to adding a phone line for $10 bucks\ month? All I wanted was a simple phone. She would not be able to use anything more complex and all she needs is a phone line period.
I have no problem paying for something that works and does what it should, but this is ridiculousness. The MTF on this device is somewhere around 1 to 2 days. Out of 365 days that's about a 96.5% failure rate....... Want my advice? Get ride of this device. It sucks and it's bad marketing.

jsavage9621,
It pains me to hear about your experience with the Home Phone Connect. This device usually works seamlessly and is a great alternative to a landline phone. It sounds like we've done our fair share of work on your account here. I'm going to go ahead and send you a Private Message so that we can access your account and review any open tickets for you. I look forward to speaking with you.
TrevorC_VZW
Follow us on Twitter @VZWSupport

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Can any super nice helpful one tell me why i have DSL,but there is no dial tone in my house?

it has been 2 weeks , at begining, my phone and DSL all down and i was been told by verizon that it' s the problem of this 's garea. ...after 2 weeks, DSL is fixed up , but there is no dial tone for my home phone. i switched another telephone, cord, still no dial tone. ...and i used my cell phone to call my home phone, dial tone is busy....
...,,right now, what can i do? i called verizon, they will send me a technician and charge me $91 for first 30mins.........i am a poor man with little english ....what can i do? feel so helpless

They will only charge you if the problem is in your residence. If you had a wire maintenance plan there should be no cost. If your DSL is working your phone should be. I would say it is not a local problem in your residence, unless the phone jack where you are pluging your phone in is bad. Try unpluging your DSL modem and pluging your phone in at that location. If it works then you have a wiring problem, and will be charged for the repair trip.
To check if the problem is in your residence.
Open the customer side of the phone box on the outside and unplug the cable from the jack.
Plug your phone in at that location, and see if it works. If it does then the problem is a wiring issue in your residence.
If your equipment coming into the house is old, you may not be able to test it this way.
Message Edited by prisaz on 05-24-2009 10:15 AM

Upgraded the firmaware for SPA-3000 and lost the dial tone

Please help me figure out the problem with my Sipura box which has lost the dial tone after I upgraded the firmware. All the configuration looks fine. it is registering fine too but no dial tone.

Reset the device to factory default. This is usually needed after upgrading the firmware.

Intermittent Dial tone-SPA-2102

Device Model: SPA-2102
Internet: ADSL Modem
WAN settings: Tried with static as well DHCP
Issue: Device lost dial tone after 2 min and got again after rebooting it.
Register Expire:60secs
NAT Mapping: Enable
At the same time able to browse the websites so it is not the issue Internet connectivity.
Can any one help me to find the root cause.
Thnx.
HM

May I ask if during the time that the dial tone is gone, did you check if the VoIP is still still having a “Registered” status? I think for the “Register Expire” to “3600 “seconds.
Try to see if changing this value will make it work. If still the same, I suggest to generate a syslog of the occurrence and contact 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

Is there a way to set up a dial tone or a way for someone who is hard of hearing to have the phone set up to make a noise when I hold it up to my ear?

Is there a way to set up an dial tone or set the phone up to make a noise when I hold it up to my ear when i use it

I'm also not sure what you want to accomplish by this. Until you hit the "call" button, the phone is not 'off hook' like a traditional landline would be. There is no such thing as a dial tone on any cell phone. The dial tone was simply a carrier signal from the central office that prepared that end to listen for the pulses (later tones) generated when you dialed.

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 & SPA2102 have too short dial tone time.

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