similar to: SIP/NAT disconnection issue

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Well, a bit of a newbie using Asterisk and trying to get some Sipura SPA-841 phones to talk to the Asterisk server. Not doing something right apparently.... If I turn on debug for the extension in the CLI (sip debug ip xxx.xxx.xxx.xxx) I see frequent 5 packet attempts by the server to contact the phone, but seems to always be failing. The status

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the

Hi, I have a little situation here :( Perhaps somebody can give me a hand with it. I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor

Hi, I have an asterisk server sitting behind a pfsense firewall, I have successfully configured pfsense for NAT traversal, and clients from the internet can call clients inside the network of asterisk, as well as other clients registered with this asterisk server on the internet. The problem now is when a client from the internet do a call, the call disconnects in 10~20 seconds, but during

I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line,

Hi people, I have a few SPA-942 around, all of them work fine except one. The one behind NAT.. In every phone you can: * Pickup a Call on one of the line buttons, * Create a new call on another button * Press "xferLx" to join those to calls. This works everywhere except on the one behind NAT. After a lot of messing around with all the options possible I gave up and subscribed

Hi, I have a little situation here :( I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor my router have DynamicDNS. So, if I

I was just reading the release notes for the latest SPA-841 firmware, and noticed that Sipura added support for "SIP-B" to this release. This apparently adds support for bridged line appearances, parking softkeys, called party ID, external missed call summary support, and a handful of other useful features. The release notes are available at

Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus

Anyone having problems with registration to * from a SPA-841? I got a SPA-841 a week ago. I noticed that sometime it could not be reached (dialed to) and it can't dial. In this case the line LED is yellow. I enabled logging to syslog and there is a hint as to what happens. For some reason sometimes it gets "401 Unauthorized" Any ideas what is happening and how to fix it? Phone

Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. One feature the SPA-841 has, which I can't figure out how to implement on the snom 190, is the "make/accept calls without registration" feature. Or more specifically, "produce a dial tone even if I'm not registered." I would like to set our

I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed

On Tuesday 10 May 2005 9:45 am, David Masure wrote: > Hi folks ! > > I bought two sipura 841 phones. I used to have GN Netcom headset which > I connect instead of the handset. The problem is that I don't have any > sound coming out the headset and I can't speak neither ! > ... > > Or....can anyone advise me on headset working with the sipura 841 ? I just use a

Hello, i've recently configured my asterisk for internal sip calls. while testing, i noticed that 1 out of 10 calls works.. at first i thought my router dropping packets around the way as it were a bottle neck.. so i've added a switch. once i tested again same prob occurs... im using xlite as a softphone on clients pc and centos server on a dedicated machine. at times the phone call

Hi list, I have a couple of questions related to asterisk billing and the generation of cdr logs. I've searched the wiki but have not found my answers, hopefully you guys can help. 1) When are asterisk CDR logs _normally_ generated? When the call arrives, when the call hangs up, or both? I have looked at the records created and it seems to only generate it at the time the call is

I am trying to get a SPA-3000 to work behind NAT - for the sake of the exercice. The SPA is on the local network at the address 192.168.0.125 behind a NATted linux router. The machine I am trying to work with is a friend's (let's call it lolo.dyndns.org) and I've installed Asterisk 1.0.3 on it. I can see the SPA register but when I try to make an outbound call I get the message:

Has anyone gotten call-waiting to function on the 4 line SPA-841? I've seen some documents that say it can do it, some say no way. If yes, can you share configs / SPA-841 settings? If no, did you work around it? I can call out just fine on all 4 lines, however, if I am on the line, another call coming in does not ring the 2nd line...it just goes to busy / VM. -Darren

Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN<->VoIP settings. Sipura Configuration website form field names are numeral only. :( [1]