Hi,<br><br>I am trying to setup a simple home voip service w/ *<br>I have compiled and installed the svn source<br>as a first step I am trying to configure SIP for inside my network.<br>I have a handful of softphones and a few hardphones that I want to all be able to call each other
<br>I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo<br>from the asterisk install (ext 500), both give me a 401 Unauthorized error<br>below I have included some debugging output and all the important config files
<br><br>*******part of extensions.conf that was added by asterisk-gui (svn)*******<br>[asterisk_guitools]<br>exten = executecommand,1,System(${command})<br>exten = executecommand,n,Hangup()<br>exten = record_vmenu,1,Answer
<br>exten = record_vmenu,n,Playback(vm-intro)<br>exten = record_vmenu,n,Record(${var1})<br>exten = record_vmenu,n,Playback(vm-saved)<br>exten = record_vmenu,n,Playback(vm-goodbye)<br>exten = record_vmenu,n,Hangup<br>exten = play_file,1,Answer
<br>exten = play_file,n,Playback(${var1})<br>exten = play_file,n,Hangup<br>hasbeensetup = Y<br><br>[DID_trunk_1]<br>include = default<br><br>[numberplan-custom-1]<br>plancomment = DialPlan1<br>include = default<br>include = parkedcalls
<br><br>[timebasedrules]<br>*******part of extensions.conf that was added by asterisk-gui (svn)*******<br><br><br>*******part of users.conf that was added by asterisk-gui (svn)*******<br>[trunk_1]<br>allow = all<br>context = DID_trunk_1
<br>dialformat = ${EXTEN:1}<br>hasexten = no<br>hasiax = yes<br>hassip = no<br>host = <a href="http://iax2.fwdnet.net">iax2.fwdnet.net</a><br>port = 4569<br>registeriax = yes<br>registersip = no<br>secret = rycort4e<br>trunkname = Custom - fwd
<br>trunkstyle = customvoip<br>username = 788694<br><br>[6000]<br>callwaiting = yes<br>cid_number = 6000<br>fullname = proton<br>hasagent = yes<br>hasdirectory = no<br>hasiax = no<br>hasmanager = no<br>hassip = yes<br>hasvoicemail = yes
<br>host = dynamic<br>mailbox = 6000<br>secret = proton<br>threewaycalling = yes<br>vmsecret = 1234<br>registeriax = no<br>registersip = yes<br>canreinvite = yes<br>nat = no<br>dtmfmode = inband<br>disallow = all<br>allow = all
<br>context = numberplan-custom-1<br>*******part of users.conf that was added by asterisk-gui (svn)*******<br><br>the rest are straight from the samples that got installed at build time<br><br>*******************debugging output*************
<br>*CLI> sip show peers<br>Name/username Host Dyn Nat ACL Port Status<br>6000/6000 <a href="http://192.168.0.101">192.168.0.101</a> D 5060 Unmonitored<br>1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline]
<br><br>debugging output from calling 500<br><--- SIP read from <a href="http://192.168.0.101:5060">192.168.0.101:5060</a> ---><br>INVITE <a href="mailto:sip:500@192.168.0.102">sip:500@192.168.0.102</a> SIP/2.0<br>Via: SIP/2.0/UDP
<a href="http://192.168.0.101">192.168.0.101</a>;branch=z9hG4bK78B64BD<br>CSeq: 2212 INVITE<br>To: <<a href="mailto:sip:500@192.168.0.102">sip:500@192.168.0.102</a>><br>Content-Type: application/sdp<br>From: "6000" <
<a href="mailto:sip:6000@192.168.0.102">sip:6000@192.168.0.102</a>>;tag=327F7192<br>Call-ID: <a href="mailto:2096168429@192.168.0.101">2096168429@192.168.0.101</a><br>Subject: <a href="mailto:sip:6000@192.168.0.102">sip:6000@192.168.0.102
</a><br>Content-Length: 230<br>User-Agent: kphone/4.2<br>Contact: "6000" <<a href="mailto:sip:6000@192.168.0.101">sip:6000@192.168.0.101</a>;transport=udp><br><br>v=0<br>o=username 0 0 IN IP4 <a href="http://192.168.0.101">
192.168.0.101</a><br>s=The Funky Flow<br>c=IN IP4 <a href="http://192.168.0.101">192.168.0.101</a><br>t=0 0<br>m=audio 33322 RTP/AVP 0 97 8 3<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:97 iLBC/8000
<br>a=fmtp:97 mode=30<br><br><-------------><br>--- (11 headers 11 lines) ---<br> == Using TOS bits 0<br> == Using CoS mark 5<br>Sending to <a href="http://192.168.0.101">192.168.0.101</a> : 5060 (no NAT)<br>Using INVITE request as basis request -
<a href="mailto:2096168429@192.168.0.101">2096168429@192.168.0.101</a><br>No user '6000' in SIP users list<br>Found peer '6000' for '6000' from <a href="http://192.168.0.101:5060">192.168.0.101:5060
</a><br><br><--- Reliably Transmitting (no NAT) to <a href="http://192.168.0.101:5060">192.168.0.101:5060</a> ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP <a href="http://192.168.0.101">192.168.0.101</a>;branch=z9hG4bK78B64BD;received=
<a href="http://192.168.0.101">192.168.0.101</a><br>From: "6000" <<a href="mailto:sip:6000@192.168.0.102">sip:6000@192.168.0.102</a>>;tag=327F7192<br>To: <<a href="mailto:sip:500@192.168.0.102">sip:500@192.168.0.102
</a>>;tag=as6b3f431e<br>Call-ID: <a href="mailto:2096168429@192.168.0.101">2096168429@192.168.0.101</a><br>CSeq: 2212 INVITE<br>User-Agent: Asterisk PBX SVN-trunk-r81159<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Supported: replaces<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f450cef"<br>Content-Length: 0<br><br><br><------------><br>Scheduling destruction of SIP dialog '<a href="mailto:2096168429@192.168.0.101">
2096168429@192.168.0.101</a>' in 32000 ms (Method: INVITE)<br><br><--- SIP read from <a href="http://192.168.0.101:5060">192.168.0.101:5060</a> ---><br>ACK <a href="mailto:sip:500@192.168.0.102">sip:500@192.168.0.102
</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.0.101">192.168.0.101</a>;branch=z9hG4bK78B64BD<br>CSeq: 2212 ACK<br>To: <<a href="mailto:sip:500@192.168.0.102">sip:500@192.168.0.102</a>>;tag=as6b3f431e<br>From: "6000" <
<a href="mailto:sip:6000@192.168.0.102">sip:6000@192.168.0.102</a>>;tag=327F7192<br>Call-ID: <a href="mailto:2096168429@192.168.0.101">2096168429@192.168.0.101</a><br>Content-Length: 0<br>User-Agent: kphone/4.2<br>Contact: "6000" <
<a href="mailto:sip:6000@192.168.0.101">sip:6000@192.168.0.101</a>;transport=udp><br><br><br><-------------><br>--- (9 headers 0 lines) ---<br>Really destroying SIP dialog '<a href="mailto:2096168429@192.168.0.101">
2096168429@192.168.0.101</a>' Method: ACK<br><br>*******************debugging output*************<br><br><br>thanks in advanced<br>Ryan<br>