[asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

[Barry Fawthrop]

what is the sip.conf for   1239
which I'm going to assume is a extension on the TNT

Barry

JR Richardson wrote:
> Hi All,
>
> I have a lab setup with two asterisk servers and a MAX TNT in the
> middle like this:
>
> asterisk sip >< sip TNT pri >< pri asterisk
>
> The TNT is running 11.0.6 and the asterisk servers are running
> 1.2.9.1.  I can get calls to pass from asterisk sip to tnt to pri to
> asterisk but not the other way. The call from asterisk to pri to tnt
> is good, the TNT is passing SIP invite to the SIP Asterisk server. I
> have tried many variations of using sip options insecure,
> autocreatepeer, permit/deny, host, user, etc.... but can't seem to get
> asterisk to accept an unauthenticated call from the TNT using SIP.  I
> keep getting SIP/2.0 407 Proxy Authentication Required.  I know others
> have done this, but with older Asterisk versions, I'm wondering what
> versions of Asterisk are known to work with the MAX TNT and with what
> version of the TNT?
>
> I'm confident this is an asterisk issue, with insecure=very, I should
> be able to pass calls to asterisk without trying to authenticate it
> first.  But this is not happening.
>
> Here is a debug of a call and a snip from my sip.conf:
>
> sip.conf
>
> [maxtnt]
> type=friend
> host=10.10.14.131
> insecure=very
> dtmfmode=inband
> callerid="MaxTNT" <maxtnt>
> context=trunktntin
> qualify=yes
> reinvite=no
> canreinvite=no
> disallow=all
> allow=ulaw
>
> debug
>
> lab1*CLI>
> <-- SIP read from 10.10.14.131:5060:
> INVITE sip:2145551212 at 10.10.14.121:5060;user=phone SIP/2.0
> t:   <sip:2145551212 at 10.10.14.121:5060;user=phone>
> f: "NO CID NAME"
> <sip:1239 at 10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a
> Remote-Party-Id: "NO CID NAME"
> <sip:1239 at 10.10.14.131:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off 
>
> i: 3a8884d9-64-1fb1f65c at 10.10.14.131
> CSeq: 639089 INVITE
> v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
> Max-Forwards: 70
> m: <sip:1239 at 10.10.14.131:5060;user=phone>
> k: replaces
> c: application/sdp
> Accept: application/sdp
> Accept-Encoding:
> Accept-Language: en
> User-Agent: Lucent-Universal-Gateway
> l: 232
>
> v=0
> o=t1gw01 531756636 531756636 IN IP4 10.10.14.131
> s=Session SDP
> c=IN IP4 10.10.14.131
> t=0 0
> m=audio 40198 RTP/AVP 0 96
> a=silenceSupp:on
> a=ecan:b on g168
> a=ptime:20
> a=rtpmap:96 telephone-event/8000
> a=rtpmap:0 PCMU/8000
>
> --- (16 headers 11 lines) ---
> Using INVITE request as basis request - 3a8884d9-64-1fb1f65c at 10.10.14.131
> Sending to 10.10.14.131 : 5060 (non-NAT)
> Reliably Transmitting (no NAT) to 10.10.14.131:5060:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131
> From: "NO CID NAME"
> <sip:1239 at 10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a
> To: <sip:2145551212 at 10.10.14.121:5060;user=phone>;tag=as41f8454e
> Call-ID: 3a8884d9-64-1fb1f65c at 10.10.14.131
> CSeq: 639089 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", 
> nonce="3ea7e98a"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call '3a8884d9-64-1fb1f65c at 10.10.14.131' in 
> 15000 ms
> Found user '1239'
> lab1*CLI>
> <-- SIP read from 10.10.14.131:5060:
> ACK sip:2145551212 at 10.10.14.121:5060;user=phone SIP/2.0
> t:   <sip:2145551212 at 10.10.14.121:5060;user=phone>;tag=as41f8454e
> f: "NO CID NAME"
> <sip:1239 at 10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a
> i: 3a8884d9-64-1fb1f65c at 10.10.14.131
> CSeq: 639089 ACK
> v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
> Max-Forwards: 70
> User-Agent: Lucent-Universal-Gateway
> l: 0
>
>
> Any guidance will be much appreciated.
>
> Thanks.
>
> JR
>