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.

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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

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

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

SPA 3102 Port Forwarding

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.

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

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

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.

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.

SPA 3102 Possible MTU issue

I have a customer using the SPA3102 that is having issues connecting to netbanking sites, swap to another router and the service works fine (except the customer looses VoIP) this screams to me MTU issue,
Is their somewhere in the SPA 3102 that I can drop the MTU size for a PPPoE session?

there's no MTU setting on the SPA3102...since there is no port triggering on the SPA3102 as well try either DMZ or port forwarding (for port forwarding just forward port 443 which is for secured sites) -- this is under Router> Application tab
| isolate! isolate! isolate! |

I have an iPhone 4 and 5 with 2 different numbers but same Apple ID and I can't set up voicemail on the 5

i have 2 iPhones with 2 different numbers but same Apple ID, why can't I set up voicemail on one phone

Visual voicemail is a carrier feature, contact your cell phone provider and see if it is set up for you to use.

Cant set up voicemail on new phone

I am trying to set up voicemail on my new iphone 5 and I am not given a prompt or any way to set it up. I reset my settings and still no luck.

I'm seeing the same thing and I can't seem to find any information on this particular problem. All of my PC Jabber installs work fine.

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 : Setting up Voicemail

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