search for: verysecret

...ed out. [2008/02/13 17:47:07, 100] smbd/chgpasswd.c:expect(279) expect: expected [*New*Password*] received [ Retype New Password:] match yes [2008/02/13 17:47:07, 10] smbd/chgpasswd.c:expect(290) expect: returning True [2008/02/13 17:47:07, 100] smbd/chgpasswd.c:expect(242) expect: sending [VerySecret ] [2008/02/13 17:47:10, 10] lib/util_sock.c:read_socket_with_timeout(476) read_socket_with_timeout: timeout read. select timed out. [2008/02/13 17:47:10, 100] smbd/chgpasswd.c:expect(279) expect: expected [*Retype*New*Password*] received [ Mismatch; try again, EOF to quit. New Password:]...

...busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty simple for these extensions; here is what I am using now: [extension1234] mailbox=1234 at default type=friend context=users host=dynamic secret=verysecret I have looked at the sample sip.conf and did not get any clues, also the SPA900 Admin Manual doesn't say anything about it. Thanks Cassius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101124/4b...

I've just seen this post: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is set to use DTMF 'outband' any ideas? Dominique -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167...

...is not my issue (I think). Here is some info: 1. * server and clients are all on the same subnet but are separated from the internet by a proxy/firewall which forces all port 80 traffic through the proxy. 2. The server has a single channel fxo card. 3. Snip of sip.conf: [test] type=friend secret=verysecret regexten=1234 ; When they register, create extension 1234 callerid="Test Unit" <1234> host=dynamic ; This device needs to register nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to...

..._plugins = " quota sieve" } protocol imap { mail_plugins = " quota imap_quota" } # Working LDAP configuration # /etc/dovecot/dovecot-ldap.conf.ext uris = ldap://he01-auth-01.greenhills-it.co.uk dn = uid=dovecot,ou=people,ou=SRV_Accounts,dc=greenhills-it,dc=co,dc=uk dnpass = VerySecret sasl_bind = no auth_bind = yes ldap_version = 3 base = dc=greenhills-it,dc=co,dc=uk scope = subtree user_attrs = homeDirectory=home,uidNumber=uid,gidNumber=gid,gosaMailQuota=quota_rule=*:storage=%$M user_filter = (|(uid=%u)(mail=%u)(gosaMailAlternateAddress=%u)) pass_attrs = uid=user,userPassword=...

...on state as well, this affects the "SASL" extension. This is a CVS version that is a couple of days old (the old 0.99.10 wouldn't run on Linux 2.6 because it has tighter resource limiting): +OK dovecot ready. capa +OK CAPA TOP USER UIDL RESP-CODES SASL PLAIN . user SECRET +OK pass VERYSECRET +OK Logged in. capa +OK CAPA TOP USER UIDL RESP-CODES . quit +OK Logging out. -- Matthias Andree

...f problems. Without resorting to OpenSER or other proxies (as my provider also uses IAX), is there a way I can load balance outgoing channels in Asterisk? For example an IAX peer like: [iax_provider] type=peer username=myprovider host=xxx.xxx.xxx.10 host=xxx.xxx.xxx.11 host=xxx.xxx.xxx.12 secret=verysecret disalow=all allow=g729 Is there any way I can balance calls between all of the hosts in the providers description? In fact, if I set the dialplan like: exten => _X.,n,Dial(IAX2/iax_provider/${EXTEN} what IP addres will receive the call? host 10, 11 or 12? I know DAHDI can balance outgoing c...

I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or