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

[Enrique Martinez]

AFAIK, in 1.2.x the insecure=very change in favor to insecure=port,invite,

also you can try with allowguest=yes

Regards

JR Richardson wrote:
> Update,
>
> I loaded asterisk 1.0.10 and it worked straight away. I can send
> unauthenticated calls to asterisk.  Something in 1.2.9.1 and 1.2.13
> are not allowing unauthenticated calls when insecure=very is set in
> sip.conf, either in the global or peer context.
>
> Are there any switches in the Asterisk Makefile to allow this?
>
> JR
>
> On 11/7/06, JR Richardson <jmr.richardson at gmail.com> 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
>>
>> -- 
>> JR Richardson
>> Engineering for the Masses
>>
>
>