similar to: SPA-2102 sending local IP instead of WAN IP in SIP packets

Topology: analog_phone-----SPA2102-----Navini_Wireless_Router------ISP------Asterisk A ping against the asterisk server shows aprox 145ms roundtrip. 128kbps upstream 512kbps downstream g729a as codec signal quality of the navini router: 100% The ATA operates correctly in every form, however sometimes when someone is talking to me (the other person is at pstn) and then I start talking the other

Been having a problem with my two Sipura 2000's dropping calls from the SPA-2000 side. Seems the calls are dropped right before the "Next Registration" time. Calls drop about ever 60 minutes or so. I have dialed from one port to the other and let it sit. After about 60 minutes or so the calls get dropped. System details are below Asterisk ver. CVS-HEAD-11/27/04-23:42:45 RHEL 3

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server was working. Doesn't fix it. I reset one of

https://bugzilla.mindrot.org/show_bug.cgi?id=2102 Bug ID: 2102 Summary: [PATCH] Specify PAM Service name in sshd_config Classification: Unclassified Product: Portable OpenSSH Version: 6.2p1 Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: PAM support

Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

I'm proposing to install an Asterisk PBX at a collocation facility for a remote customer. Each of the customer locations will have an SPA-3000 with the FXO port connecting a POTS circuit and the FXS port connecting a fax machine or red phone. In addition to voice traffic, the customer has a high volume of incoming and outgoing faxes. Would it be possible, using g711 between the SPA-3000

Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a "normal" phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very

Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think of it because the Sipura 2100 is a broadband router, the voice part may be considered as being behind

Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater was sent as "maxBitRate". Without capital "M". Are you closer to

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

In my hunt to track down my echo issues, I tried disabling all echo cancellation, suppression, adaption, on my SPA-2000 (Advanced section of the config, under Line 1/2). Then calling from one local extension to another. (SPA-2000 Line1, to Line2 on the same device) I was pretty shocked with the results, the echo was HORRIBLE! I even tried 3 different analog phones. Now, once I turned the echo

My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk "mainmenu" context (or ext I guess). Currently the

Greetings, I'm having a heck of a time with one way audio on a SPA2012. It's public IP connected directly to cable modem. One line configured. Asterisk is multihomed Public IP outside / Private Inside. Extensions inside network are can't hear audio from phone outside connected via the spa-2012. Outside can here audio from inside the network. Ring works both ways. I've

Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? I'm a bit confused as to where my SPA-2000 is currently getting its time. I told it GMT-5 in the misc section but it doesn't really tell me where its going for this. Is it just broadcasting looking for ntp? The net of my problem is that it is 1 hour

I recently lost my setup (bad drive) and I'm now trying to get my setup back. I have Asterisk setup to a BT100, a Cisco 7960 (7.2 SIP) and an SPA-3000. I can call the phone extension, I can call from the phone on the SPA to other extensions and I can call out to the PSTN. What I can't do is to call from the PSTN to through the SPA to Asterisk. It rings twice then I get fast busy. I

hello my friend is trying to get his spa-1001 (sipura) 1001 connected to my asterisk box. he reset his spa-1001 to factory defaults I emailed him the voip-info page I found on google and yes I did look on google anyways he isn't able to get the thing to connect to it eg getting a dial tone, he did install x-lite and it worked fine with that am running asterisk@home 1.0 can some one please tell