Re: [Asterisk-Users] Compiling * on OpenBSD 3.5

2004-08-01 Thread mpwspam-digiumlist
Thanks for that. It certainly did need that. When I tried again I got undefined refs to sin, cos etc.. I wonder why it compiled fine the last time for me.. only difference is that I have the sources in /usr/src - but that shouldn't be required I think.. No I am getting an undefined ref to

Re: [Asterisk-Users] Compiling * on OpenBSD 3.5

2004-07-31 Thread mpwspam-digiumlist
Fantastic - Many thanks! For the purposes of the archive, this is what I did.. Edited /usr/src/asterisk/Makefile Just after:- ifeq (${OSARCH},Darwin)LIBS+=-lresolvendififeq (${OSARCH},FreeBSD)LIBS+=-lcryptoendifLIBS+=-lssl I added:- ifeq (${OSARCH},OpenBSD)LIBS=-lcrypto -lpthreadendif And it

[Asterisk-Users] Compiling * on OpenBSD 3.5

2004-07-30 Thread mpwspam-digiumlist
Hi, Has anyone had any success? After a clean install of OpenBSD, I do the following:- pkg_add ftp://rt.fm/pub/OpenBSD/3.5/packages/i386/gmake-3.80.tgz pkg_add ftp://rt.fm/pub/OpenBSD/3.5/packages/i386/bison-1.35p1.tgz pkg_add ftp://rt.fm/pub/OpenBSD/3.5/packages/i386/ruby-ncurses-0.8.tgz Then,

Re: [Asterisk-Users] Re: New Beta version of Grandstream Firmware 1.0.5.9

2004-07-27 Thread mpwspam-digiumlist
Not sure - I emailed their support and they told me to either:- a) contact my service provider for the update or b) set the tftp server IP address to 4.3.153.50 - but check with my service provider first.. Anyway - as my own service provider I gave myself permission and set the TFTP IP on the

Re: [Asterisk-Users] IAX dialing indication tone

2004-07-26 Thread mpwspam-digiumlist
I'm a bit of a noob at this myself - and don't know for sure - but try changing the tm (at the end) to tmr.. The r should proving a ring tone to the calling party.. Michael.spkao [EMAIL PROTECTED] wrote: Hi,I am using the following to dial a far end IAX server from a local IAXserver bothusing

RE: [Asterisk-Users] X100P Inbound Issue

2004-07-26 Thread mpwspam-digiumlist
Hi, I had already come across some of that stuff (forgot to post that part of my sip.conf). Here is what I'm using right now:- FROM SIP.CONF GENERAL disallow = allallow=ULAWallow=ALAWallow=GSM canreinvite=no [001] ; Budgetonedisallow = allallow=GSMallow=ULAWallow=ALAWallow=ilbc ... The

RE: [Asterisk-Users] X100P Inbound Issue

2004-07-26 Thread mpwspam-digiumlist
Dave, Thanks for that - I had missed that one. It dosn't make any difference to the problem though - SIP calls and outbound PSTN work fine - inbound PSTN causes this very strange problem.. Michael.Dave Cotton [EMAIL PROTECTED] wrote: On Mon, 2004-07-26 at 04:10 -0700, wrote: Hi, I had already

[Asterisk-Users] X100P Inbound Issue

2004-07-25 Thread mpwspam-digiumlist
Hello, After much searching of voip-info.org and google, I'm finally giving in and asking the list. The setup I have is this:- Single X100P card in a Debian system Inbound/Outbound POTS line connects to the X100P Sipura 2000 and Budgetone 100 on the LAN 1 Cordless and one conventional phone