Discussion: Sipura 3000

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United States Regional Settings

Before starting, I returned my SPA3000 to factory defaults. Now that everything is working, I have an odd problem with inbound phone rings. it is not the typical steady US phone ring. It's three rings of very short durations. I have looked everywhere, but can't seem to find a good example of United States regional settings.

Here's what I have:
Ringback Tone: 440@-19,480@-19;*(2/4)
Cadence: 60(2/4)

If someone could screenshot or post US settings for the entire advanced Regional page and help me out?

by cjkline83, Wednesday 24 of October, 2012 (22:28:04 UTC)
HK new world telecom (NetTalk) PAP2 config

HongKong new world telecom is offering VOIP service at HKD470 per year with a hongkong local tel number If you call HK a lot or you have families / friends in HK who call you a lot, it is worth giving it a try.
I just registered and no credit card info is required.

Now my question: currently it is only offering a softphone option. I have a Linksys PAP2 SPA3102, does anyone know how to config it for HK new world telecom (NetTalk)? here is the website for HK Nettalk: http://www.nwtbb.com.hk/ct_index.html

please tell me in step by step, thanks


by laputat99, Sunday 05 of August, 2007 (10:40:44 UTC)
BroadTel RPA-2E1S1O

BroadTel RPA-2E1S1O is a better alternative and the best buy in the market. It supports Tone configuration including dial tone, ring tone, ring back....and so on. It even allows registration to 3 SIP proxies. and many more.....Check it out on www.broad-tel.com

by newbie, Monday 19 of June, 2006 (02:43:12 UTC)
Thanks for the guide...

Hello Everyone! .. I have followed this guide and now got my Sipura3000 to work for the incoming call now. I have been working on this for several weeks and finaly found this guide and it is working for me now... but now I need to setup my Sipura3000 so that it can do an outgoing call from asterisk ... if anyone here got it to work please post the setup here....

thanks...

by overseacalling, Wednesday 22 of February, 2006 (12:48:41 UTC)
dtmfmode=rfc2833 appears broken on spa-3000

A comment in http://voip-forum.tmcnet.com/voip-forum/forum/forum_posts.asp?TID=1707&PN=1&TPN=1 indicates that the SPA-3000 doesn't appear to support dtmfmode=rfc2833 since revision 2.0.10 of the firmware. I can confirm this for the 2.0.13 firmware too. (But dtmfmode=inband works). The symptom is that DTMF (after dialing) don't get passed through the PSTN port to the remote side, making IVR with a bank impossible.

by jbd, Sunday 22 of January, 2006 (17:50:51 UTC)
Re: Echo problems on FXO port

While working on another installation, I've found echo can still be a nightmare to tune out on the SPA 3k. From forums on Voxilla.com, I've found a lot of reports that the 3.1.7 firmare is plagued with echo problem. Most of the same reports said 3.1.3 is a better release when echo is concerned.

Still I've found playing around with the line impedance is necessary in most cases to get it to work. Here in the USA the default setting of 600 should work, but on 6 different lines I've placed it now, 900 has worked better.

The PSTN to SPA gain adjustment seems to have almost no effect. If I can find a tool to analyize the audio charateristics of the SIP stream or within asterisk it would certainly help in getting these things dialed in.

I'd be currious to know who's had great success with echo control with these and what firmware version they are using.

by lschweiss, Thursday 19 of January, 2006 (15:00:08 UTC)
RTP Packet Size

By default the SPA 3000 has the RTP packet size set to 0.030. This causes some poor sound qualities at times coming from Asterisk sound files. Change this to 0.020 and sound quality is excellent.

by lschweiss, Saturday 10 of December, 2005 (20:34:12 UTC)
Echo problems on FXO port

I've just installed two SPA 3000's with their FXO ports connected to POTS lines and Asterisk terminating the SIP connection with local SIP phones. However, I'm seeing consistant echo problems when connected to cell networks, but not landline based callers. If a plain phone is connected to these lines no echo problem is there. I have set jitter buffer the the minium and echo cancel is on. Any sugestions?

UPDATE: After an extensive reading on echo cancelization in Asterisk I figured out that if you reduced the SPA to PSTN gain the echo will go away. I set it to -4 and the echo is gone. Apparently the volume of the outgoing sound was creating more echo than the echo canceller could handle in the SPA.

by lschweiss, Saturday 10 of December, 2005 (17:57:13 UTC)
Using the PSTN to dial out

The Sipura works great as an external FXO device. BUT unfortunately if you use it to dial out, it connects first the voip call and than dials the number on the FXO port. So the time connected will start right away and it is not possible for asterisk to try to call a different number on busy or on no answer (since the voip call is successfully established!!) :-(

by doerner, Saturday 15 of October, 2005 (06:51:13 UTC)
Re: Call waiting on the PSTN line

I am having the same exact problem ... did u have any luck getting this issue resolved?

by shorki, Thursday 29 of September, 2005 (02:32:36 UTC)