Hi Ryan,
If you look at the [numberplan-custom-1] section in extensions.conf,
you'll discover that it doesn't include 'demo', so that's likely why
that part doesn't work. Add
include => demo to that section, and do an asterisk -rx 'extension
reload' from the command prompt.
As for local extension dialing -- that's got me a bit baffled. I can't
figure out where you're suppose to enable that in the AsteriskNOW GUI.
Anyone know how to active that? Are they using the realtime dialplan
lookup, or enum or something?
On 8/29/07, Ryan Murray <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I am trying to setup a simple home voip service w/ *
> I have compiled and installed the svn source
> as a first step I am trying to configure SIP for inside my network.
> I have a handful of softphones and a few hardphones that I want to all be
> able to call each other
> I have configured users.conf with a single softphone(kphone) and have tried
> calling itself (ext 6000) and the demo
> from the asterisk install (ext 500), both give me a 401 Unauthorized error
> below I have included some debugging output and all the important config
> files
>
> *******part of extensions.conf that was added by asterisk-gui (svn)*******
> [asterisk_guitools]
> exten = executecommand,1,System(${command})
> exten = executecommand,n,Hangup()
> exten = record_vmenu,1,Answer
> exten = record_vmenu,n,Playback(vm-intro)
> exten = record_vmenu,n,Record(${var1})
> exten = record_vmenu,n,Playback(vm-saved)
> exten = record_vmenu,n,Playback(vm-goodbye)
> exten = record_vmenu,n,Hangup
> exten = play_file,1,Answer
> exten = play_file,n,Playback(${var1})
> exten = play_file,n,Hangup
> hasbeensetup = Y
>
> [DID_trunk_1]
> include = default
>
> [numberplan-custom-1]
> plancomment = DialPlan1
> include = default
> include = parkedcalls
>
> [timebasedrules]
> *******part of extensions.conf that was added by asterisk-gui (svn)*******
>
>
> *******part of users.conf that was added by asterisk-gui (svn)*******
> [trunk_1]
> allow = all
> context = DID_trunk_1
> dialformat = ${EXTEN:1}
> hasexten = no
> hasiax = yes
> hassip = no
> host = iax2.fwdnet.net
> port = 4569
> registeriax = yes
> registersip = no
> secret = rycort4e
> trunkname = Custom - fwd
> trunkstyle = customvoip
> username = 788694
>
> [6000]
> callwaiting = yes
> cid_number = 6000
> fullname = proton
> hasagent = yes
> hasdirectory = no
> hasiax = no
> hasmanager = no
> hassip = yes
> hasvoicemail = yes
> host = dynamic
> mailbox = 6000
> secret = proton
> threewaycalling = yes
> vmsecret = 1234
> registeriax = no
> registersip = yes
> canreinvite = yes
> nat = no
> dtmfmode = inband
> disallow = all
> allow = all
> context = numberplan-custom-1
> *******part of users.conf that was added by asterisk-gui (svn)*******
>
> the rest are straight from the samples that got installed at build time
>
> *******************debugging output*************
> *CLI> sip show peers
> Name/username Host Dyn Nat ACL Port Status
> 6000/6000 192.168.0.101 D 5060 Unmonitored
> 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0
> offline]
>
> debugging output from calling 500
> <--- SIP read from 192.168.0.101:5060 --->
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK78B64BD
> CSeq: 2212 INVITE
> To: <sip:[EMAIL PROTECTED]>
> Content-Type: application/sdp
> From: "6000" < sip:[EMAIL PROTECTED]>;tag=327F7192
> Call-ID: [EMAIL PROTECTED]
> Subject: sip:[EMAIL PROTECTED]
> Content-Length: 230
> User-Agent: kphone/4.2
> Contact: "6000" <sip:[EMAIL PROTECTED];transport=udp>
>
> v=0
> o=username 0 0 IN IP4 192.168.0.101
> s=The Funky Flow
> c=IN IP4 192.168.0.101
> t=0 0
> m=audio 33322 RTP/AVP 0 97 8 3
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
>
> <------------->
> --- (11 headers 11 lines) ---
> == Using TOS bits 0
> == Using CoS mark 5
> Sending to 192.168.0.101 : 5060 (no NAT)
> Using INVITE request as basis request - [EMAIL PROTECTED]
> No user '6000' in SIP users list
> Found peer '6000' for '6000' from 192.168.0.101:5060
>
> <--- Reliably Transmitting (no NAT) to 192.168.0.101:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK78B64BD;received= 192.168.0.101
> From: "6000" <sip:[EMAIL PROTECTED]>;tag=327F7192
> To: <sip:[EMAIL PROTECTED] >;tag=as6b3f431e
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2212 INVITE
> User-Agent: Asterisk PBX SVN-trunk-r81159
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f450cef"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog ' [EMAIL PROTECTED]' in 32000 ms
> (Method: INVITE)
>
> <--- SIP read from 192.168.0.101:5060 --->
> ACK sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK78B64BD
> CSeq: 2212 ACK
> To: <sip:[EMAIL PROTECTED]>;tag=as6b3f431e
> From: "6000" < sip:[EMAIL PROTECTED]>;tag=327F7192
> Call-ID: [EMAIL PROTECTED]
> Content-Length: 0
> User-Agent: kphone/4.2
> Contact: "6000" < sip:[EMAIL PROTECTED];transport=udp>
>
>
> <------------->
> --- (9 headers 0 lines) ---
> Really destroying SIP dialog ' [EMAIL PROTECTED]' Method: ACK
>
> *******************debugging output*************
>
>
> thanks in advanced
> Ryan
>
--
| It ain't what you don't know that gets you into trouble. It's what
| you know for sure that just ain't so. -- Mark Twain
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