Folkert van Heusden
2005-Oct-30 13:00 UTC
[Asterisk-Users] dialout gives 404 (using sjphone (dialin works fine))
Hi, I'm running asterisk 1.0.9 with an FXO card. People can call me on my pstn line and that gets transferred to my laptop (on 192.168.62.100). That all runs fine. If, though, I want to dial out (a pstn line) I always get a "call rejected: 404 not found" error (in sjphone) and in the asterisk console this appears: Check for res for is not a local user is not a local user Stopping retransmission on '66ED2B84-1DD2-11B2-8B72-D132D0F7B1FD@192.168.62.100' of Response 1: Found In sip.conf I have this: [1000] type=peer host=dynamic defaultip=192.168.62.100 dtmfmode=rfc2833 mailbox=0000 context=dialoutcont callerid="Folkert van Heusden" <folkert@keetweej.vanheusden.com> and in extensions.conf: [dialoutcont] exten => _0XXXXXXXXX,1,Dial(ZAP/1/${EXTEN}) Anyone got a suggestion what might be wrong? Folkert van Heusden -- Try MultiTail! Multiple windows with logfiles, filtered with regular expressions, colored output, etc. etc. www.vanheusden.com/multitail/ ---------------------------------------------------------------------- Get your PGP/GPG key signed at www.biglumber.com! ---------------------------------------------------------------------- Phone: +31-6-41278122, PGP-key: 1F28D8AE, www.vanheusden.com -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 282 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051030/e8bf240d/attachment.pgp