similar to: Sipura SPA2000 & Asterisk & latest firmware (1.0.18)

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___________ HOME _______________ ____OFFICE ____ SPA2000 <---> Linux Box <--> Asterisk Box 192.168.0.253 192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention

Hi I have just got Asterisk going with an spa-2000. however when I look through the userpdf every function on the sipura and asterisk seems to require on-hook or flash button , all of the phones i have do net seem to have either, is there a way round this ? does anyone know. Or do i ahve to go out and buy more phones? Anyhelp appreciated Simon

I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver

I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2 on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share the same IP address. Every minute I repeatedly get the following output: SIP Debugging Enabled 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6 SIP/2.0 Via:

Not I. -----Original Message----- From: Kevin [mailto:Asterisk@gtcus.com] Sent: Tuesday, June 01, 2004 7:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sipura-SPA2000 background noise I have been using Cisco ATA's for analog connections and decided to give a Sipura SPA-2000 a try. I noticed there is a fair amount of background white noise that is noticeable, especially

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I have to make a config change to any of these

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

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

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

Anybody ever run into a case where the Sipura Dial Plan will not work with the S0 option to immediately connect? My Dial plan reads (*xx|[3469]11S0|0|00|[2-9]xxxxxxS0|1xxx[2-9]xxxxxxS0) and I can dial ONLY then numbers in the dial plan so I know that it works. For some reason when I dial 5551212 1212121212 It does not dial for a while and then it dials 555 1212 Anyone have any ideas?

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER

Hi all! I am frustrated. I am new to asterisk. My system is ASTLINUX if receive a Fax on my sipura spa2000 i get this: Feb 25 07:41:00 NOTICE[1708]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060225/ca251876/attachment.htm

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse

Some people on this group may have understood from messages posted here that the Sipura SPA-2000 is not currently available for shipping. That is not the case. Voxilla.com has the Sipura SPA-2000 available for immediate shipping, and has had them since late November. The price is $109.95, and it comes with a month of free VoicePulse service with activation fees waived (a $65 value). In return

Dear Asterisk Users, I have a Sipura 2000 device, and since last few days I have been searching for its latest firmware for upgrade. Googling tells me that Cisco has stopped the support for this device and I dont have definite idea on where would I be able to find the firmware to upgrade my device. Any help in regards to getting the firmware will be helpful. Regards, Amol -------------- next part

Dear All ~ I have * setup & running ok (with two Wildcard X100P's to PSTN). I also have two analog phones connected into same through a SIPURA 2000. These work fine, except that when I call out through PSTN & try to send DTMF tones to (say) a remote PBX to dial an extension, the gain seems to go wild (high), and the DTMF tones are not recognized at the other end. I tried setting the

Hi everyone, Im having some troubles getting work Asterisk as SIP Client and a Huawei softswitch as SIP server. I already got my asterisk registered to the Huawei. Im working with a Sipura SPA 2000 registered to Asterisk. When im trying to make an incoming call from the Huawei to asterisk it rings but when i answered the call drp down inmediatly. The sip debug finally show this