similar to: OT Old Sipura OK - Linksys (junk)

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sorry, I am resending this, I tried earlier, but I couldn't see it appear on the archives - apologogies if it appears double! -------------------------------------------------- My Sipura 3000 ATA died on me this morning. I had a Linksys SPA 3102 available which I would like to use as a replacement. Unfortunately, the SPA3102 is not able to register with the asterisk server - I am

Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

Will the DAHDI with OSLEC have any effect one ECHO in Linksys or Sipura adapters? I know Dahdi is a replacement for Zaptel. And Zaptel is mostly for internal cards not external units like Linksys. I have one Linksys 3K that with so echo so bad that it is pretty much useless. -- Joseph

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

Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura -- #Joseph GPG KeyID: ED0E1FB7

I have a Linksys Sipura SPA2102 connected to Asterisk 1.4.27 and sometimes it doesn't connect at all. I keep getting a busy signal when I try to dial. It appears to happen most often when both lines are registered. The 2 lines on Linksys lines also use different ports. Does that mean than it is necessary to configure the 2nd line with port=5061 in sip.conf, or can they both use 5060? Thanks

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

First of all, I'm not knocking Sipura/Linksys. I have heard very good things about their products. I'm just wondering if they are the only quality shop on the market. I know about the zoom 5801 where you can't dial out the FXO from SIP, only from the FXS port. And I have heard similar about the HT-488 also. I want to know if anyone else makes ATAs where all of the features work

Hello, I have found asterisk is dialing a random number when it recieves a call, would anyone know why? The first thing I noticed found peer 4563 (this is a n Xlite Client) Many thanks, Spencer SIP Debugging Enabled spitfire*CLI> <-- SIP read from 82.70.154.145:5060: INVITE sip:448715046363@iptel.tgfslp.dalmany.co.uk SIP/2.0 Max-Forwards: 10 Record-Route:

I'm going to be deploying around 30 IP phones with Asterisk in the near future. I've tentatively settled on the Linksys/Sipura SPA9xx family. I am unclear on the notion of "lines" in the context of SIP phones like these. The SPA942 model has a 2-line-to-4-line upgrade available, but I don't know why I'd need to purchase it. I have tested a SPA942 with Asterisk, and

I'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret=1111

All, I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file,

Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Through our reseller, we couldn't get any clue to Linkys-Sipura products error codes. This keeps us from analysing Syslogs. Has anyone a clue to get this error codes and their meaning ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061017/eb5e3bbe/attachment.htm

I was just browsing my local suppliers list of headsets - not a single one with a single 2.5mm jack. Either USB or 1-2 3.5mm jacks. Can anyone recommend a headset that works with e.g. SPA-921 and -941? /Per Jessen, Z?rich

Hello Gurus I've installed my Asterisk server for testing on the company I work the setup or the approach let's call it is: 1 Asterisk Server fully configured and with some SIP extensions setup on two cities A and B. 2. One local PSTN line connected thru a x01p card to call local phone numbers numbres on city A. 3. A Sipura 3102 Gateway on city B connected to a city's B PSTN line. I

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear