similar to: SPA 2000 and 404 not found

For some reason, I can't get call pickup to work between Sip phones or between Sip and Zap phones. All phones are in the same call group and pickup group (1). The source code was downloaded and built as of today 11/15/03. Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=aliens ; ; SIP Entry for sipura line 1 ; This

Hello all, I am trying to configure a subdomain DNS server on a Cent OS 5.1 - for my lab. Brief configuration: Lab machine ---> 192.168.17.2 (should respond to DNS queries from hosts in 192.168.16.0/20 network) 1. I would also like to forward any queries outside the above network to our corporate domain (no firewalls between our 192.168.16.x network and the corporate network. The domain

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

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

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

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

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'm getting frustrated attempting to understand; I googled and asked folks and am unable to get a straight answer. 1. How is the /etc/resolv.conf file maintained ? I do not seem to get a consistent result when I save resolv.conf configuration from GUI or by hand using vim /etc/resolv.conf. a. Sometimes I the entries toggles between the two entries: # generated by

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

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! I have many problems getting this thing to work. There''s a host with two network interfaces, where there are two routers to Internet in two separated networks. The host uses multihop routing for deciding to which router send the packets... but the routing decision is wrong made. Some packets with source address of one NIC, go to other network. I have a host with three NICs in it:

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'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

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 MP3 files and going

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

I am blessed with this user forum and able to set my Dev-PCI Digium card working fine with the Asterisk server of mine (i)But today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance --------------

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

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

All, Here's a cool one.. I was attempting to call a retarded conferencing service, and was having problems with it picking up my DTMF.. after trying all the settings my Sipura SPA2000 offers, I found inband actually works.. unfortunately, I can't get anything else to pick up my inband DTMF (including asterisk's builtin voicemail! It just times out and says I never entered a login!).