Spa9k spa400 and pap2t

I have these items (9k, 400, pap2t) working together, however, when calling a spa voip phone (spa942) from a phone attached to the pap2t, I am unable to end up in voicemail if the user at the spa942 does not answer. From other SPA phones it is not a problem. I know there are issues with generic SIP clients getting into voicemail, but does that include the scenario I describe above? Do any of the spaxxx ATA's other than the pap2 have the ability to get forwarded into VM if an extension does not answer? Thanks.

I wanted to post a reply from another forum to what I had asked here. Looks like Cisco says ATA's won't be able to leave VM. You had indicated it should be possible....have you seen it working? Does the explanation below make any sense? Is there a different configuration that might work?
1. Feb 27, 2009 4:50 PM in response to:
Re: spa9k spa400 and pap2t
Hi,
I simulated your environment in my lab and ran a sniffer to capture and analyze the network traffic prior to answering here.
The issue is that the SPA9000, SPA400, SPA9xx phones, and WIP310 phones are a tightly coupled solution.
Non-SPA/WIP310 phones and ATAs do not supply the correct information in their SIP INVITE to the SPA9000 when attempting to leave vmail for the SPA942 on the SPA400.
SPA/WIP310 phones include a P-Mailbox: <mbox-ID>
in the SIP INVITE to the SPA9000 when attempting to leave vmail. [sip:vm@<SPA9000_IP>:6060]
In contrast, the PAP2T's analog phone sends a SIP INVITE to the SPA9000 but does not include the P-Mailbox: <mbox-ID> information because the PAP2T, nor any other ATAs are designed to work in the SPA9000 Voice System solution.
In summary, because the PAP2T's SIP INVITE to the SPA9000 to sip:vm@<SPA9000_IP>:6060lacks P-Mailbox: <mbox-ID> information, the SPA9000 responds with a 503 Service Unavailable and the analog phone caller finally hears a busy tone when the call is dropped.
Regards,
Patrick

Similar Messages

Problem with spa400 and pap2t

Hello,
I have a problem with spa400 and pap2t. If from analogue telephone behind pap2t call outside, work, cal call go out go into spa400 e on the rtg system.
If I call from external, the calll arrive to spa400 but the spa400 don't forward the call to papa2t.
Suggestions?
Thanks

No the experiment is just with spa400 and pap2t. spa 9000 there isn't.
There is a method for using the call incoming and outcoming only with this spa400 and pap2t.
Thanks!

Uc560 and pap2t configuration

Hi all,
we have a UC560 with t1 and we would like to connect some analog dect to it (we already haveSPA504 connected and working fine).
for the analog dect we would like to use linksys pap2t
do someone know how to configure it?
Thanks
Diego

Just so I'm clear, you want the PAP2T to use the UC520 as a SIP provider e.g. Analog phone hard wired to PAP2T, SIP to UC520, Unknown to PSTN.
Can you post your PAP2T config? It should be a simple SIP registration from the PAP2T to the UC520, much like any SIP phone.
Tip easy way to manage your PAP2T config is SPA Conf, I can't find it online anymore so here is a direct link to my copy
For Mac/Linux
https://www.dropbox.com/s/qh8sy3utvyvy9iv/spaconf-0.3.tar.gz
For Windows
https://www.dropbox.com/s/q0mekobskrntjnt/spaconf-0.3.zip
MC

PAP2t EU Problem - Dialing Duplicate Numbers

Hello,
Guys, i have PAP2T EU which creating the problem like dialing the duplicate numbers. For example I dial 025023456 (typical brussels number) it will dial 022550002344555 somthing like that. What to do with it?
Please help me out to sort out this problem.
Regards
Salman Ahmad

Hello,
the problem may also be incompatibility in the impedance between the attached phone and FXS port impedance setup.
Also, the phone may generate tones on the edge of norm / out of norm / and PAP2T may realy "hear" every tone twice.
So please check with another phone, check the impedance, and - last but not least - make sure you only use 2wire cable
(inner 2 wires only) between PAP2T LINE1 socket and your phone - certain SPA-2102 and PAP2T models has both lines
wired on LINE1 socket and it may cause incompatibility with the attached phone if 4 wire cable is used.

Trixbox SPA400 disconnect PSTN line after 60 seconds

Hi all, I just bought a SPA400 and am trying to connect it to my Trixbox. I followed the HOW TO guide in the voxilla forum, and in no time I was able to make outbound calls via the SPA400. This is my trunk set up
allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=192.169.0.71
insecure=very
nat=no
type=friend
user=SPA400
However, almost all of my calls got disconnect 60 seconds into the calls. I captured the packets, and it was the SPA400 that first sent the RTPC Goodbye message to my Trixbox.
Right after I power cycled the SPA400, things worked well for about 5 minutes or so.. then the hang ups. Does anyone seeing this problem? Can you please help me?
I am running the latest Trixbox, and had tried different SPA400 versions ranging from 1.0.0.5 to 1.0.0.12. Thanks!

I am having the same problem, very random though.
SPA400 disconnect PSTN line after 60 or more seconds Outbound Calls Only it seems.
I had the same problem with trixbox 1.2.
spa400 Firmware Version: 1.0.0.12
Netgear-FS750T2-Switch Managed with QoS Enabled between the Server and the SPA400
Phones Grandstream GXP2020 and GXP2000
OS Version: Linux asterisk.berkleigh.com 2.6.18-8.1.15.el5
#1 SMP Mon Oct 22 08:32:04 EDT 2007 i686 i686 i386 GNU/Linux
Asterisk Build: Asterisk 1.4.13 Asterisk GUI-version Revision: 1752
No indication of problem at asterisk console #asterisk -rvvvvv.
I am now monitoring the SPA400 Logs.
Things I see in Logs that I do not understand.
These are not in order and extracted from different parts of the logs. 2235862- SSMU 4 *** ERROR -- DOES not INCLUDE CNUM
3842- SSMU 4 SSMU: 3, ssmup_proc_connect attempted
40592- NMM 5 NMM: set_dccfg() calls tiuo_fxo_set_curlimit(1)
605610972- NMM 4 msntp: 0, too many bad or lost packets
I am debating weather to try the Digium TDM400P we already have ( but had echo problems on early asterisk versions ) or buy a Grandstream GXW4104 4 port FXO Gateway.
Message Edited by berkleigh on 11-30-2007 12:37 PM
Message Edited by berkleigh on 11-30-2007 12:40 PM

SPA9000/SPA400 Initial Setup Problems

Hi All
I hope someone out there will be able to help a newbie to VOIP as we're getting nowhere slowly and painfully with the documentation. Our setup is:
SPA9000 connected to SRW208P switch via the Internet port on the SPA9000
SPA400 connected to the SPA9000 via the switch above
SPA400 connected to a BT (British Telecom) POTS socket on port #1.
2 x Linksys SPA922 phones connected to the switch
1 x Ordinary analog phone connected to FXO Port #1 on the SPA9000
At the moment we've got no SIP service connected and hope to use the system entirely on POTS for now
So far we've run the wizard and sucessfully connected everything together, assigned IP addresses etc. The SPA9000 can "see" the SPA400 and we've triple checked the addresses, users and ports match up OK. Likewise the phones are connected and we've been able to configure them and we can make internal calls on the system OK. During the wizard configuration we've set Line #1 on the SPA9000 to the attached SPA400 and assigned 9 to choose that line.
The problems start when we try to make or receive a call on the external line via the SPA400. When trying to make a call (either dialling the number directly or prefixing with 9) then we get an engaged type sound and it doesn't ring out. On the SPA400 management page all four lines are showing as on-hook although line 1 is showing a voltage etc. The line 1 status light is also flashing very quickly at all times. The BT line is known to work and has nothing else connected to it. For incoming calls the system doesn't pick up the line and it just keeps ringing.
Finally the analog phone connected to the FXO #1 port on the SPA9000 appears completely dead with no dial tone and you can't ring out internally or externally using this phone.
We've tried numerous combinations of configuration but nothing seems to make any difference. Some advice would be very appreciated if anyone can spare the time for a quick how to guide.
Many Thanks
Andy.

Andy,
I've had the same issue with a SPA9000/SPA400 install on a BT line in the UK. The FXO port LED would blink continuously and the line would appear dead - no pickup on incoming calls, and outbound calls would show service not available on the phone.
After a *lot* of experimentation, I managed to get the line stable enough to get this working, but its a workaround and not a solution.
If you syslog'ed the SPA400 you got entries such as:
Sep 14 21:51:44 192.168.0.14 0000715174 - TIU: 3 FXO voltage changed, state on, voltage 7V -> 0V^M
Sep 14 21:51:46 192.168.0.14 0000716210 - TIU: 3 FXO voltage changed, state na, voltage 0V -> 24V^M
Sep 14 21:51:47 192.168.0.14 0000716805 - TIU: 3 FXO voltage changed, state on, voltage 24V -> -5V^M
Sep 14 21:51:48 192.168.0.14 0000717304 - TIU: 3 FXO voltage changed, state on, voltage -5V -> 8V^M
Sep 14 21:51:49 192.168.0.14 0000717805 - TIU: 3 FXO voltage changed, state on, voltage 8V -> 14V^M
Sep 14 21:51:50 192.168.0.14 0000718305 - TIU: 3 FXO voltage changed, state on, voltage 14V -> 20V^M
Sep 14 21:51:51 192.168.0.14 0000718804 - TIU: 3 FXO voltage changed, state on, voltage 20V -> 23V^M
and you can see the line voltage changing continuously.
So, I added a splitter to the line at the master socket, and put a known working standard phone on the same line as the SPA400. So, this provided a stable impedance to the line, and the line settled at -47V and no voltage fluctuations.
Given that I am using 320 Ohms + (1050Ohms || 230nF) which is Linksys's suggested value, and I have tried 370 Ohms + (620 Ohms || 310nF) as shown in this BT Document (http://www.sinet.bt.com/351v4p5.pdf), and the default of 600Ohms all of which resulted in an unstable line im confused as to what BT are doing to the line.
Andy, does the incoming PSTN line you are using have ADSL on it? or did you manage to find a real solution?

PAP2T with Axon server proble

Hi all,
I've been using Axon together with X-Lite and have had no issues with it. Recently, I bought a Linksys PAP2T... Unfortunately I've been having lots of trouble with it.
The PAP2T, Axon server and X-Lite are all on the same subnet with no firewalls/routing/etc.
- When I call the PAP2T from X-Lite, the call is successful. However, when I hook the phone (connected to the PAP2T), Axon does not drop the call (and X-Lite still shows the call as active).
- When I call X-Lite from the PAP2T, the call connects for around 8 seconds (sound passes through normally) and then suddenly the call is dropped.
- When I call from Phone 1 to Phone 2 of the PAP2T (both registered as different extensions on Axon), the call only connects for about 8 seconds (as above) after which the call drops.
Does anyone have any idea what might be wrong? It's been quite frustrating for me and I've dug around many forums but cannot find any answers unfortunately.
Thanks!
Regards,
Tim.

Hi, I have the SPA9000 as my PBX, new Gigaset phones for users and PAP2T for connecting older non voip phone (polycom conf phones)
The whole setup is on the same LAN but behind a NAT/Firewall with DHCP coming from the NAT/Firewall.
The SPA9000 is connected to my network via the WAN port
I want both the VOIP phones and the PAP2T to use the SPA9000 as its SIP provider.
I have now got the PAP2T working with the SPA9000!! If it helps anybody else this is my setup:
I have a password set on the SPA9000 (set by going to SIP tab on the SPA9000 (from admin/advanced login mode) and setting Phone Ext Password: - which is right at the bottom).
NOTE: You may not need to set a password - I do because I'm also using using Gigaset VOIP phones that wont connect to the SPA9000 without a password.
On the PAP2T I have set (using the web interface of the PAP2T) from the Line1 TAB
a) Sip port to be one not used by any other phone (unique)
b) Proxy to the ip address and port of my SPA9000 (e.g. 192.168.1.2:6060) . You will need to set the values based on your setup of the SPA9000
c) Set user ID to the extension no I wanted the phone to be on (e.g. 2503)
d) Set the password to be the same as the SPA9000
e) Set the use Auth ID to YES
f) Set the Auth ID to the extension no I wanted the phone to be on (e.g. 2503)
g) Line enabled to yes!
And it works!
Thanxs
s
Message Edited by siman007 on 12-01-2008 10:48 AM

Nee Help and Suggestions

HI Linksys Admin I am planning on wiring my house with cat 6 or 7. Witch cat is batter to use. I have one VOIP line, two Stander line, one ADSL line. With this I went to put a switchboard or intercom system so I can call any room in the house, transfer calls, answer the door when someone comes to the door. What the best why to go about it and what produces should I use. Can you show it in a diagram so I can see what it look like. Take care

take a look at an SPA9000/ / SPA400 and SPA942/SPA962 combo's
just not sure about the intercom or how your switch board is setup but the device should prettty much handle you internal calling function.

PAP2T Wireless Conflict with WRT610 Router

I have a WRT610 Router with a PAP2T for my VOIP. I have 5 PCs on the network. Three wired and 2 wireless. One of the wireless PCs looses connection when I dial out on the VOIP The ips are not the same and the MAC addresses are diferent.
Any thoughts?

Basically PAP2T is a phone adapter and run on DHCP...And there might be an IP Address conflict between your one computer and PAP2T...So, you may try giving the static ip to the computer....Go to start...control pannel...network connection and then right click on wireless network connection and click on properties...highlight "internet protocol tcp/ip" and click on properties.Choose use the following IP Address and put the ip address 192.168.1.10 and the default gateway 192.168.1.1(assuming your router is 192.168.1.1) and click on OK...It should work.

PAP2T is Droppiing DIal Tone after around 1 hour when connected to D-Link DSL-504G

I have recently changed my router to a D-Link DSL-504G and now my PAP2T looses dial tone randomly but usually after 15 -60 mins of use. It never comes back on unless I reset the router and PAP2T. Then it does the same.
I have an external static IP address (ISP) I have a static IP address for my router. I have made a internal static IP address for the PAP2T. I have port forwarded as per my SIP providers instructions (maybe have made an error here?).
I have called D-Link and they don't have a clue. I have upgraded the D-Link to the latest firmware. I have NOT upgraded the PAP2T as it used to work wit h my other router. (Not available anymore)
Is this a configuration problem on the D-Link or is it just incompatible?

Hello,
if you have forwarded SIP ports (5060 and 5061 by default) make sure you also forward the RTP port range (16384-16482). The loss of the dialtone appears to be tight to loss of the registration - which can be related to dead/wrong conntrack entry somewhere on the way between you and your VoIP provider.
Next time you loose the tone try to change "SIP Port" on the line in trouble from 5060 (5061) to something (exotix) else, like 6050 (6051).
You may also consider to completely cancel the port forwarding and use the NAT keeplive ( $INFO or $NOTIFY) every 5 or so minutes (300s) to keep the NAT alive between the $REGISTER requests (which are every 3600s by default AFAIK)

Call park using fxs port on spa9000

I have seen this question asked several times but no seems have answered it.
Is it possible to enter a service code to park a call using an analogue phone. If it is, please share what is the service code and the procedure.
I have a spa9000 with pstn (spa3002, spa400) and exts spa-942,xlite and spa2102/pap2. Everything works fine on the 942 phones and the xlite softphones but cant get more that the *98 transfer feature working on the analog phones.

Not so sure if the service code being utilize by the FXS is the same as what is mentioned in VSA.Then again, the manual says "dial the corresponding * code on the client station" and if you look at an SPA9xx phone, it has the following codes defined for call park and call unpark code.
*38
*39

SPA9000 for Newbie

Hi all, I am thinking of buying the SPA400 and SPA900 for my home business. I have never used a PBX before. There are a few questions I would like to ask:
1) We have only one PSTN line. Do we need at least two lines/phone numbers in order to forward an extension to an outside number? That is if an incoming call is directed to an extension which is set to forward to my mobile phone, do we need two phone lines to do that? One for incoming calls and one for the forwarding?
2) If my line has call waiting and a second caller calls in, can the SPA9000 pick up the second call at the same time the first call is still connecting?
3) I am confused by the terms multi-line hunting and DID. Where can I get more information about these terms?
Any information will be appreciated! Thank you!

i believe that all the configuration that you need will work with the SPA-9000 and SPA-400. i suggest that you visit the FAQ for SPA-9000 so you can verify if it suit your needs.

SPA9000 - transferring calls to my home

Hello.
Background:
My accountancy practice is about to install spa9000 and spa400 and linksys phones.
I would like to be able to make a call from home to the office and vice versa.
At home I have an ASDL (broadband) service in New Zealand. I use a wireless router( a netgear dg834gt) at home.
I am happy to change the router if needed.
Question:
Will this type of router work, or, do I need one with special protocals. Does linksys have a suitable one?
Is it as simple as this or is there some other equipment I need.
Thankyou
Mark

The router is fine as long as you have internet. All you have to do is plug the WAN port of the SPA-9000 to the LAN port of the router. Connect the SPA-400 and IP phones to any available LAN ports on the router. If you are using the Linksys Voice System IP phones then it will automatically register to the SPA-9000. All you need to think is the VOIP provider to be registered on any Lines of the SPA-9000.

SPA 9000 Config

Is it possible to config the SPA 9000 to work in this environment?
Depending on what line is being called in to go to a AA message. For example someone calls in using Line1 it gets AA message A and if someone else calls in on Line2 and they will get AA message B? I have a client who has 2 businesses located in the same building and is trying to use the same Voice system but keeping them seperate. They do have a total of 2 SPA 9000's and 2 SPA 400's. I currently have it setup only using 1 9000, and 2 400s (each has 1 analog line coming in and only 1 is configured for voicemail for all extensions).
Any Ideas?
Thanks,
Matt

No for what it is my understanding. AA settings are system wide so just one settings for any SPA9000. But, anyway, if you can have the PSTN line on different SPA400 and each line belongs to the different company, you can route them to the different SPA9000 and rustle up the game.
Max

Dialing problem

i have an asterisk server, and i have made an number such as 10* to dial and then i can set up some features on the asterisk.
my cisco phone 7960 is doing it fine, but my wrp400 and pap2t don´t.
any help ?
Solved!
Go to Solution.

Lunatic, and what exactly hapens when you dial 10* and wait ?
Do you get any dialtone at beginning, any beep or tone during or after dialing 10*, any tone after waiting let's say 10-15s (after you dial 10*) ?
Does the Asterisk indicate any way that "feature set" mode is activated ?
The dialplan of PAP2T shall include one of those like :
|xx.|
(if you use "random" two digits)
|1x.|
(if you use numbers 10*-19*)
|10*S0|
(if you always dial 10* only and you want it to get sent immediatelly)

Spa9k spa400 and pap2t

Similar Messages

Maybe you are looking for