Hi all,
I’ve done a test install of asterisk 1.2.22 on openbsd 4.2. After some troubles I managed to register to my VOIP service provider.
sip show registry shows I’m registered.
However, with the extensions.conf file I’m having much more difficulties.
My extensions.conf file looks like this:
; $OpenBSD: extensions.conf.sample,v 1.1.1.1 2004/09/26 00:38:24 jolan Exp $
[default]
exten => _0[1234567]XXXXXX,1,Dial(SIP/tiscali/${EXTEN},60,tr)
exten => _0[6].XXX,1,Dial(SIP/tiscali/${EXTEN},60,tr)
exten => s,1,Answer
exten => s,n,Ringing
exten => s,n,Dial(SIP/tiscali,20,tr)
exten => s,n,Wait,1
exten => s,n,Congestion
When I ring my phone, the CLI gives:
Nov 26 20:44:36 WARNING[32687]: file.c:517 ast_openstream_full: File welcome does not exist in any format
Nov 26 20:44:36 WARNING[32687]: file.c:828 ast_streamfile: Unable to open welcome (format alaw): No such file or directory
Nov 26 20:44:36 WARNING[32687]: pbx.c:5826 pbx_builtin_background: ast_streamfile failed on SIP/172.26.30.3-7fef4000 for welcome
Nov 26 20:44:36 WARNING[32687]: file.c:517 ast_openstream_full: File enter-ext-of-person does not exist in any format
Nov 26 20:44:36 WARNING[32687]: file.c:828 ast_streamfile: Unable to open enter-ext-of-person (format alaw): No such file or directory
Nov 26 20:44:36 WARNING[32687]: pbx.c:5826 pbx_builtin_background: ast_streamfile failed on SIP/172.26.30.3-7fef4000 for enter-ext-of-person
Nov 26 20:44:36 WARNING[32687]: file.c:517 ast_openstream_full: File or-press does not exist in any format
Nov 26 20:44:36 WARNING[32687]: file.c:828 ast_streamfile: Unable to open or-press (format alaw): No such file or directory
Nov 26 20:44:36 WARNING[32687]: pbx.c:5826 pbx_builtin_background: ast_streamfile failed on SIP/172.26.30.3-7fef4000 for or-press
Nov 26 20:44:36 WARNING[32687]: file.c:517 ast_openstream_full: File to-reach-operator does not exist in any format
Nov 26 20:44:36 WARNING[32687]: file.c:828 ast_streamfile: Unable to open to-reach-operator (format alaw): No such file or directory
Nov 26 20:44:36 WARNING[32687]: pbx.c:5826 pbx_builtin_background: ast_streamfile failed on SIP/172.26.30.3-7fef4000 for to-reach-operator
Nov 26 20:44:46 WARNING[32687]: pbx.c:2435 __ast_pbx_run: Timeout, but no rule ‘t’ in context ‘default’
Nov 26 20:44:56 WARNING[32687]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission 15c37b70-5748f0c3-13c4-242146-29c012f2-242146@172.26.30.3 for seqno 1 (Critical Response)
And I see that the server picks up the phone, but I hear nothing.
What I’m expecting is that I haven’t installed all of asterisk. Can’t seem to find the sounds directory anywhere. Is it that my installation is not complete? I’ve installed it from the package provided by OpenBsd and I just did a pkg_add asterisk.
Any help is appreciated.
