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This looks significant: root@pbx:~ $ java -version java version "1.4.2" gij (GNU libgcj) version 4.1.2 20080704 (Red Hat 4.1.2-48) Copyright (C) 2006 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for...

Yep....... Although, I did try this a year or so ago (with your very original instructions), so had already got Java installed. How would I start again?

root@pbx:/siptosis $ ./SipToSis_linux Launching SipToSis Exception in thread "main" java.lang.NoClassDefFoundError: local.ua.SkypeUA at java.lang.Class.initializeClass(libgcj.so.7rh) Caused by: java.lang.ClassNotFoundException: java.util.concurrent.TimeoutException not found in...

I had a good go with this last week and it looked much more flexible than v1. However, I couldn't get it to provision any phones.........was I trying a little too early before the release? I downgraded to v1 but will try again now it's been released for test.

have a look at this. http://www.voip-info.org/wiki/view/Using+Slimserver+for+Music+on+Hold

Well, commenting it out doesn't stop it from working, so it would seem, that it does nothing.....

So, in order to get the time working, it looks like this needs to be inserted.....but where? Set(DB(lastcalltime/204)=${EPOCH})

Thanks for your reply. Do you mean like this? [custom-app-1871] exten => s,1,Answer exten => s,n,Wait(1) exten => s,n,Macro(user-callerid,) exten => s,n,NoOP(${DB(lastcalltime/204)) exten => s,n,SayUnixTime(${DB(lastcalltime/204)},Europe/London,q \'digits/at\' IMp)...

OK, so I worked out the bit that needed changing (with help from someone on uk.telecom.voip): [custom-app-trace] exten => s,1,Answer exten => s,n,Wait(1) exten => s,n,Macro(user-callerid,) exten => s,n,SayUnixTime(${DB(lastcalltime/204)},Europe/London,q \'digits/at\' IMp) exten =>...

So, I think I need to modify the code below. [custom-app-trace] exten => s,1,Answer exten => s,n,Wait(1) exten => s,n,Macro(user-callerid,) exten => s,n,Playback(info-about-last-call&telephone-number) exten => s,n,Set(lastcaller=${DB(CALLTRACE/${AMPUSER})}) exten => s,n,GotoIf($[ $[...

Hello, I have an extension, 204 which forwards to my cell. Because none of my ITSPs allow CLI to be forwarded, should I miss a call, I'd like to be able to dial in on a DID which points to DISA, be authenticated and then dial *69 to discover the CLI of the last caller. Where should I start...

Brilliant! Brilliant! It now works with Inbound Routes. How did I miss that? Thanks.

No, I simply have (<S0:1>) as the destination in the default DP for PSTN to VoIP Gateway. (1 is a ring group on PBX) So I'm told! I've tried the following on many occasions: (S0<:xxxxxx>) (S0<:[email protected]:5060>) where xxxxxxx matches an inbound route What seems to happen is, rather...

What phones are you using? I'm seeing the same issue on an Asterisk server (not PiaF) with Snom phones

Hello, I have for many years now been using a number of SPA3000s as analogue trunks. Originally with almost every AAH incarnation and, so far, every release of PiaF (Thanks for PiaF, btw) Anyway, I have never been able to get the SPAs to come into the PBX using Inbound Routes; I always have to...

Perhaps I am tired.....I don't understand what you're asking...:confused5:

What prevents anyone you call from pressing # and getting confused by initiating a transfer?

You might be right. I shall try that tomorrow....will have to set everything static...but, no matter. Thanks.

Thanks, Joe. Yes, I've had to disable iptables; I do have a firewall/router in place, however, I've got a couple of remote extensions so have had to open up relevant ports on the router. I would like to have Iptables running too.....for the benefits of fail2ban.

Hello, I am stuggling to configure Iptables to allow any and all traffic from devices connected to the LAN. I have media servers running on the PiAF machine (we have very few simultaneous calls, so the hardware copes perfectly) I find that with Iptables enabled, the media server, although...