search for: spa2000

Displaying 20 results from an estimated 45 matches for "spa2000".

2003 Dec 10
4
Sipura SPA2000 & Asterisk & latest firmware (1.0.18)
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* s...
2005 Sep 30
2
analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly
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 before dialing numb...
2003 Nov 12
1
SPA 2000 and 404 not found
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,...
2005 Jul 02
1
Sipura SPA2000 behind NAT
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.xx...
2004 May 30
4
Sipura-spa2000
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
2004 Jun 01
0
Sipura-SPA2000 background noise
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 after breaking the dial tone. After pressing a '1' to break the dial tone, there...
2004 Jul 01
2
IAX2 to IAX2 connection problems
...Sip read: SIP/2.0 100 Trying To: <sip:2200@192.168.1.103> From: "asterisk" <sip:asterisk@192.168.1.100:0>;tag=as17b86f84 Call-ID: 56ed32e50261bf093fd0be0a6a31b8ac@192.168.1.100 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.100:0;branch=z9hG4bK64eb8c95 Server: Sipura/SPA2000-1.0.33 Content-Length: 0 8 headers, 0 lines...
2004 Nov 29
2
SPA-2000 Dropped calls
...0;branch=z9hG4bK-b1938413 From: 8445985 <sip:8445985@192.168.0.5>;tag=c864004bd9b6bbbdo0 To: 8445985 <sip:8445985@192.168.0.5> Call-ID: 76662903-a6afea65@192.168.0.20 CSeq: 1 REGISTER Max-Forwards: 70 Contact: 8445985 <sip:8445985@192.168.0.20:5060>;expires=9999 User-Agent: Sipura/SPA2000-2.0.11(g) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 12 headers, 0 lines Using latest request as basis request Sending to 192.168.0.20 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG...
2003 Sep 28
3
FYI-New ATA clone out
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 http://www.grandstream.com/y-product.htm
2005 Mar 21
1
Net2Phone / Vonage
...Sip read: SIP/2.0 200 OK To: <sip:bedroom2@192.168.5.143:5061> From: "asterisk" <sip:asterisk@192.168.5.200>;tag=as5feb5ba4 Call-ID: 5b0c2dae5d7fd21b06b6e3c822cc4050@192.168.5.200 CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK5be6f565;rport Server: Sipura/SPA2000-1.0.30 Content-Length: 0 8 headers, 0 lines Destroying call '5b0c2dae5d7fd21b06b6e3c822cc4050@192.168.5.200' 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:bedroom1@192.168.5.143:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK1fc74fe3;rport From: "asterisk&quot...
2004 Jan 04
8
Grandstream Handytone 286 RTP Problems
...ndshakes, but when * sends handytone RTP packets, I see a ICMP Port Unreachable messages sent from Handytone to * regarding the UDP RTP packet. * then gives up and I see a BYE from *, which handytone acks. Handytone config is default except obvious SIP registration parameters. I also have a Sipura SPA2000 and everything works perfect for that one, same extension and everything (not at same time of course). sip.conf entry: disallow=all ; Disallow all codecs allow=ilbc allow=ulaw ; Allow codecs in order of preference [131] type=friend host=dynamic reinvite=no...
2003 Dec 12
1
Chagres Technologies _WHERE IS MY ORDER?
Hello...Sorry for posting this here.but I can't see any other way to get a hold of JOHN BROWN I placed an order of 4 - SPA2000's with Chagres Technologies over 2 months ago. John what is the status of my order? I have emailed, faxed and called..but still no reply from you or your company. If you cannot get the product, a refund would be in-order... Thanks for your help! Darren -------------- next part...
2006 Feb 25
2
Unknown RTP codec 100 received
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
2007 Mar 01
1
Tesco Internet Phone
...net 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 see if I could get this to work. Julian.
2004 Apr 08
1
Live Music on Hold
I have a small * system in my home (1 U100S, 1 X100P, 1 BT101, and 1 SPA2000) to handle my requirements. I would like to add Music on Hold and have been watching the forum to see if something would come across on this topic. The difference I am interested in is getting the music from a radio or someother external source. All references to MOH up to now have been using MP...
2004 Jan 05
7
RE: Inexpensive Analog Ports
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
2005 Jan 05
3
Sending DTMF to PSTN problem with SIP
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
2006 Oct 18
4
Asterisk + Huawei
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
2004 Oct 07
1
Confused about NAT and Authentication with FWD
...8.0.160:5060;branch=z9hG4bK-2729a2aa From: SPA2202 <sip:2000@192.168.0.163>;tag=d7f857520fa6972o0 To: <sip:467919@192.168.0.163> Call-ID: 4dd7bf1-74628526@192.168.0.160 CSeq: 101 INVITE Max-Forwards: 70 Contact: SPA2202 <sip:2000@192.168.0.160:5060> Expires: 240 User-Agent: Sipura/SPA2000-2.0.10(e) Content-Length: 428 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 25855460 25855460 IN IP4 192.168.0.160 s=- c=IN IP4 192.168.0.160 t=0 0 m=audio 16388 RTP/AVP 2 0 4 8 18 96 97 98 100 101 a=rtpmap:2 G726-32/8000 a...
2003 Dec 17
9
Grandstream Early Dial
I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? -------------- next part