If I receive a Phone call by ISDN or from SIP Provider, the Asterisk make some errors and the SIP Client don't react.
The messages from Asterisk in verbose mode:
er will net.
Asterisk messages in Terminalmode:
parse_srv: SRV mapped to host sip-ha.web.de, port 5060
Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown" <sip:[EMAIL PROTECTED]>;tag=as5bfdabe6
Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown" <sip:[EMAIL PROTECTED]>;tag=as76a8acb1
Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown" <sip:[EMAIL PROTECTED]>;tag=as29a2f623
Mar 10 00:02:18 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown" <sip:[EMAIL PROTECTED]>;tag=as44aed266
-- parse_srv: SRV mapped to host sip-ha.web.de, port 5060
-- creating pipe for PLCI=0x101 msn = 456456
-- started pbx on channel (callgroup=0)!
== Starting CAPI[contr1/456456]/3 at ,5480080,1 failed so falling back to exten 's'
== Starting CAPI[contr1/456456]/3 at ,s,1 still failed so falling back to context 'default'
Mar 10 00:04:42 WARNING[5776]: pbx.c:1882 ast_pbx_run: Channel 'CAPI[contr1/456456]/3' sent into invalid extension 's' in context 'default', but no invalid handler
-- Executing Hangup("CAPI[contr1/456456]/3", "") in new stack
== Spawn extension (default, h, 1) exited non-zero on 'CAPI[contr1/456456/3'
-- CAPI Hangingup
-- removed pipe for PLCI = 0x101
Here is my sip.conf: [general]
bindaddr = 0.0.0.0 port = 5060 context = default maxexpirey = 3600 defaultexpirey = 120 srvlookup = yes tos = 0x18 disallow = all allow = gsm allow = alaw allow = ulaw allow = g729
register => christoph:[EMAIL PROTECTED]/christoph.hehl
[web_de] context = default type = friend host = sip.web.de username = christoph secret = password fromuser = christoph fromdomain = sip.web.de dtmfmode = inband nat = yes insecure = no
[chris] type = friend secret = passwd host = dynamic dtmfmode = rfc2833 nat = no callerid = "chris" <11> canreinvite = no qualify = no insecure = very
my extensions.conf static = yes writeprotect = no
[globals]
[default]
exten => h,1,Hangup
exten => 11,1,Dial(SIP/chris,,tr) exten => 11,2,Hangup
exten => 456456,1,Dial(11,,tr) exten => 456456,2,Hangup
exten => _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,tr)
exten => _0.,2,Hangupexten => _1.,1,Dial(CAPI/@456456:${EXTEN:1},,tr)
exten => _1.,2,HangupPlease Help
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
