similar to: How to keep difference between 2 SIP-accounts/trunks from same server ??

Hi, I do not understand the difference between SIP and IAX, is it only two different protocols or something more special. The problem I have is that I've created two users Aix.conf register => users1:passwd1 register => user2:passwd2 [user1] type=user context=default secret=passwd1 host=dynamic [user2] type=user context=default secret=passwd2 host=dynamic extensions.conf exten

Asterisk-server 1 : register => hostedasterisk:pass1 at domain2.biz [BOX-YOCAN] type=user ; incoming from YOCAN... auth=rsa inkeys=Key ; pub key of user secret=XXX ; pass2 of YOCAN context=from-BOX-YOCAN ; incoming from YOCAN trunk=yes transfer=no qualify=yes [BOX-YOCAN] type=peer ; outgoing to YOCAN... host=dynamic ; registers auth=rsa outkey=kEy ; private secret key

Hello list ! SETUP : Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone PROBLEM : I've noticed that when I put down the horn of my Grandstream it still takes a while for my GSM/CellPhone to stop ringing. INFORMATION : This is the output on the CLI of the local Asterisk-server : [Oct 3 17:40:33]

I am totally not understanding this : My IAX.conf : register => BOX-YOCAN:passwd at remote_asterisk_ip On remote Asterisk : *CLI> [Aug 30 20:37:07] -- Registered IAX2 'BOX-YOCAN' (AUTHENTICATED) at ip:4569 So this is normal... Now the following : [remoteasterisk] type=peer host=ip remote asterisk auth=md5 secret=passwd On the remote Asterisk : [BOX-YOCAN] type=user

Hi, I have a Samba 3 server which authenticate in AD. Is it possible to map windows usergroups against UNIX ones? In the logs, I see several SIDs whenever an user authenticate itself, but I don't know how to use them. For example, here is my [global] [global] dos charset = iso-8859-1 unix charset = iso-8859-1 workgroup = INFELEC realm = REALM.LOCAL

Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:XXXX at 85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret=XXXX fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes

https://bugzilla.samba.org/show_bug.cgi?id=6151 Summary: --safe-links can be fooled by adding extra slashes to the path Product: rsync Version: 3.1.0 Platform: x64 OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

hello we have a samba server on centos 5.2 and 2 different lans. so we gave the server to ips eth0:172.16.93.217 and eth1: 192.168.89.3 but after this when we tried to join clients (windows xp) to the domain the error: "the specified domain either does not exist or could not be contacted." what is the solution?

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Hi! I use protocol manager on a boot diskete to map a network drive share which is shared by a samba server. (i.e net use i: \\server\d) I give a username and a password which exists in /etc/samba/smbpasswd (i.e user name: user1 and password: passwd1) but when I create and/or edit a file, it changes the ownership to that file to owner user: dobody and group owner: nobody. Also the rights are

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status

Is there any documentation on IAX RSA authentication because I followed http://www.voip-info.org/wiki/index.php?page=Asterisk+iax+rsa+auth and it's not working... Asterisk 1 : -r--r--r-- 1 root root 272 Aug 25 10:34 server2.pub -r-------- 1 root root 963 Aug 24 19:38 server1.key Asterisk 2 : -r-------- 1 root root 963 Aug 24 19:53 server2.key -r--r--r-- 1 root root 272 Aug 25 09:02

Problem 1 : Incoming conversations from the SIP-provider come into the [default]-context and to the 's'-extension. I am unable to change this, even if I have : sip.conf [general] ;context=default ; Default context for incoming calls register => 092779077:XXXX at 85.119.188.3 ; incoming [092779077] type=user host=85.119.188.3 context=from3starsnet So I define no

Dovecot 1.07rc1 on ubuntu 6.04 I get the following problem: root 26375 0.0 0.2 5548 2240 ? Ss Sep15 1:39 /usr/sbin/dovecot root 26376 0.0 0.3 7896 3364 ? S Sep15 0:31 \_ dovecot-auth root 26378 0.0 0.3 8304 3824 ? S Sep15 0:34 \_ dovecot-auth root 26379 0.0 0.4 8668 4152 ? S Sep15 0:38 \_ dovecot-auth root

Hi list, I'm stuck for the moment @ the following : My Query (in a macro) : exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\ vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\ SIPACCOUNT="${ARG1}") Asterisk CLI : [Aug 30 14:07:42] -- Executing [s at macro-vakantie:2] MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set

Every morning I check my SIP registry to the SIP-provider. And I must conclude that during the night somewhere registry has failed. asterisk*CLI> sip show registry Host Username Refresh State Reg.Time 85.119.188.3:5060 092779077 105 Failed Sun, 05 Jul 2009 23:11:40 asterisk*CLI> sip reload [Jul 6 10:30:43]

I'm having a little problem with voicemail. Actually I'm not getting the ability to leave a voicemail-message. This is part of the dialplan : > exten => s,n(voicemail),PlayBack(/var/lib/asterisk/sounds/voicemail/${ARG1}) > exten => s,n,NoOp(${ARG1}@boxes) > exten => s,n,Voicemail(${ARG1}@boxes) > exten => s,n,Hangup() > exten => s,n,MacroExit This is the