[asterisk-users] Authentication Issue Between Servers

[Joshua Billings]

No one ever responded to this inquiry but I figured out what the issue 
was.  I thought I would respond with the solution just in case someone 
runs into the same issue in the future.

Firstly, when setting up trunking between servers the "username =" field 
is not optional. :)  Also, I had a lot of extra fields in place that I 
didn't need but hadn't taken the time to remove.  I have developed the 
opinion that config files should be kept as lean as possible.  Here is 
the revised SIP peer configuration from sip.conf:

[trunk]
type = friend
username = trunk
callerid =
context = default
host = 172.21.235.1
secret = password
canreinvite = no
disallow = all
allow = gsm



Joshua Billings wrote:
> I've got an issue where I am trying to route calls between Asterisk 
> Servers.  I can route calls inbound to a server but seem to have an 
> authentication issue going out over the same sip account.  It appears 
> that my server isn't sending the second invite after proxy 
> authentication request.  I can't figure out why; any ideas would be 
> greatly appreciated.  Thanks!
>
> - Josh
>
>
> Here is my sip.conf:
>
> [general]
> context = default
> allowoverlap = no
> bindport = 5060
> bindaddr = 0.0.0.0
> srvlookup = yes
> externip = 172.21.235.2
> localnet = 172.21.235.2/255.255.0.0
> dtmfmode = rfc2833
> relaxdtmf = yes
> disallow = all
> allow = ulaw
> allow = gsm
> maxexpirey = 30
> defaultexpirey = 180
> relaxdtmf=yes
> canreinvite = no
> nat = 0
> UserAgent = Asterisk
> echocancel = yes
> echocancelwhenbridge = yes
> t38pt_udptl = no
>
> [trunk]
> type = friend
> callwaiting = yes
> caller id =
> contact =
> context = default
> fullname =
> group =
> hasagent = no
> hasdirectory = yes
> hasiax = no
> hasmanager = no
> hassip = yes
> host = 172.21.235.1
> secret = [password]
> threewaycalling = yes
> registersip = yes
> canreinvite = no
> nat = no
> dtmfmode = rfc2833
> registeriax = no
> disallow = all
> allow = gsm
> register=>trunk:[password]@172.21.235.1
>
>
> Here is the applicable portion of extensions.conf:
>
> [default]
> exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt)
>
>
> Here is the SIP Debug output:
>
> INVITE sip:510 at 172.21.235.1 SIP/2.0
> Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
> From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
> To: <sip:510 at 172.21.235.1>
> Contact: <sip:3874 at 172.21.235.2>
> Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
> CSeq: 102 INVITE
> User-Agent: Asterisk
> Max-Forwards: 70
> Date: Tue, 30 Jun 2009 19:09:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 239
>
> v=0
> o=root 11411 11411 IN IP4 172.21.235.2
> s=session
> c=IN IP4 172.21.235.2
> t=0 0
> m=audio 11486 RTP/AVP 3 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> ^@
> ^[[KWBPBXFG000304*CLI>
> <--- SIP read from 172.21.235.1:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 
> 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;received=172.21.235.2;rport=5060
> From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
> To: <sip:510 at 172.21.235.1>;tag=as045cd609
> Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
> CSeq: 102 INVITE
> User-Agent: Asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", 
> nonce="4c4374da"
> Content-Length: 0
>
>
> <------------->
> ^@
> ^[[KWBPBXFG000304*CLI>
> --- (11 headers 0 lines) ---
> ^@
> ^[[KWBPBXFG000304*CLI>
> Transmitting (NAT) to 172.21.235.1:5060:
> ACK sip:510 at 172.21.235.1 SIP/2.0
> Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
> From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
> To: <sip:510 at 172.21.235.1>;tag=as045cd609
> Contact: <sip:3874 at 172.21.235.2>
> Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
> CSeq: 102 ACK
> User-Agent: Asterisk
> Max-Forwards: 70
> Content-Length: 0
>
> ---
> ^@
> ^[[KWBPBXFG000304*CLI>
> [Jun 30 14:09:25] NOTICE[11434]: chan_sip.c:12253 
> handle_response_invite: ^@Failed to authenticate on INVITE to '"Marci" 
> <sip:3874 at 172.21.235.2>;tag=as5951033c'
> ^@
> ^[[KWBPBXFG000304*CLI>
> Really destroying SIP dialog 
> '430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2' Method: INVITE
> ^@
> ^[[KWBPBXFG000304*CLI>
> Really destroying SIP dialog 
> '0fe5f50f7674160d2ab3522f09060d46 at 127.0.0.1' Method: REGISTER
> ^@
> ^[[KWBPBXFG000304*CLI>
>
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