similar to: Sipura-spa2000

All, If you currently own a Sipura SPA2000, avoid going to the sipura website and upgrading the firmware. I upgraded my SPA2k a couple of days ago from 1.0.9 (what it came with) to 1.0.18 off the site, and I am having issues with my SPA rebooting itself every 3-10 minutes for no apparent reason. I have been in touch with the *excellent* sipura support folks, and they are working with me to

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

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 All, I have a very small setup of 4 users and a X100P. Asterisk is currently running on an Athlon 1800 but the server it is running on is also our imap/web/mail/development/samba server and we are having a few issues with asterisk which I believe is down to to many tasks. What I intend to do is build a box just for asterisk(Ill call the old one obelix). I have a few machines in the loft

Hi All, I have a asterisk box that is now on its own static address on the net.it was originally behind a nat firewall. The problem I have is that the remote SPA-2000's that are behind nat firewalls now fail. here is relevent sip.con entry [2001] type=friend username=2001 host=dynamic defaultip=81.178.77.67 allow=ulaw dtmfmode=rfc2833 mailbox=2001@local context=sip

Does anyone know of any inexpensive alternatives to the four port analog module offered by Digium ($305) what work seamlessly with asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/1644638e/attachment.htm

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 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:

I've gotten hold of a "Tesco Internet Phone" which is a dect phone with the base connecting to the pc via usb. Has anyone been able to get this working with any softphone like xlite ? It seems as if the tesco internet phone uses IAX - the software that comes with it is a rebranded firefly (or so it seems) I already have a SPA2000 and SPA3000 hooked up, but I was just curious to

Hi, after finally getting my testsystem to work well I'm planning to buy some hardware. Now I was wondering what the advantages / disadvantages of using a Wildcard TDM400P solution over using real sip phones like the snoms is? The only thing I came up with is that the Snoms can use a LDAP adressbook on the server. What about calling a sip adress like person@server.com. Can this be done by

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

Hi, I have two DECT cordless phones (one Philips Onis 6311 and one Philips Onis2 Memo-6511). The first one is for the french standard and the second one for the british standard. Both of them have callerid functionality. The british one does not show anything when connected to my PSTN line. I have not yet tested the french one. What I want to do is to connect both of them to the same ATA box.

Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the

Anyone have practical experience using inexpensive GAP-compliant DECT handsets with the Snom M3 basestation? When I asked Snom support, the answer was that 'basic functionality should work', but they didn't elaborate. I'm _guessing_ that means registering/unregistering with the base, making calls, and receiving calls (including presenting caller ID). They also stated that they

Dear Guys, One of my customers ask me for a Wireless IP PBX, well the Asterisk fits IP PBX, But what about Wireless, ans also portable, a very special cutomer :) 1.-one possible solution is a PDA with one of those SIp softphones but VERY EXPENSIVE for the PDA 2.- a fxs channel bank and lots of Panasonic like wireless phone but the question is 1.- can my customer use all the features

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

Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I