search for: lothlorien

Hi all, I need to route local generated packages depending on which tcp or udp service I need to use. To accomplish this I have configured two routing tables: [root at lothlorien ~]# ip ru ls 0: from all lookup 255 32762: from all fwmark 0x2 lookup FirstLan 32763: from all fwmark 0x1 lookup SecondLan 32764: from 172.25.80.10 lookup SecondLan 32765: from 172.25.70.18 lookup FirstLan 32766: from all lookup main 32767: from all lookup default My routing tables: [r...

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

...Priority: P2 Component: sshd AssignedTo: unassigned-bugs at mindrot.org ReportedBy: fv at linuxvar.it Created an attachment (id=1732) --> (https://bugzilla.mindrot.org/attachment.cgi?id=1732) patch to log comment during auth phase (from <20090804121133.GA5707 at lothlorien.passione>) Hi, Attached is a patch to log key comments upon successful login. It just adds them to the already existing verbose() call. I find it useful e.g. on shared accounts where it's sometimes not enough to have the key fingerprint in the log file. Can this be applied? -- Configu...

Hello. I'm a new Asterisk user, but I'm impressed with the flexibility and versatility of Asterisk, and am moving quickly to adopt it's main-line use in our company. Hopefully, you'll be hearing more from me as the project moves forward. Right now, though, I have a question about SIP peer registration. Right now, for our SIP-based phone,s, we're using the Sip Express Router

Hi Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. Thanks in advance. P

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. -Brian On Feb 11, 8:07am, "Juki" wrote: } Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins } Hi all, } } I have Asterisk running on FreeBSD 4.x and I have made configurations to }

Hello. I'm not sure what's going on with the gentleman who is having trouble receiving inbound calls as of this weekend, but I can say that while inbound works for me, calling out through BroadVoice doesn't work at all. SIP traces show that when I send an invite request out to BroadVoice, they send back a 401 unauthorized message which includes a WWW-Authentication: header which

I'd be curious about this as well. In Asterisk version 1.0.7, it can't possibly work, unless my C reading skill is completely broken, because the voicemail app isn't listening for a "*" but only for a "#" or a "0". That's also true of /app_voicemail.c/1.203/Thu Mar 10 19:33:15 2005//D2005.03.10.08.00.00 For those interested, I've created a patch

Hello folks. I'm not sure if this is the right list for this question, but I'll start here. If I'm using a SIP provider and I have an entry in sip.conf that looks like: [8315551212] type => friend ... dtmfmode => inband ... When I pick up the phone, call someone through this provider, and press numeric digits to generate dtmf tones, who is actually generating the tones at the