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:[[email protected]
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" <sip:[email protected]>;tag=as5951033c
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=as5951033c
To: <sip:[email protected]>;tag=as045cd609
Call-ID: [email protected]
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" <sip:[email protected]>;tag=as5951033c
To: <sip:[email protected]>;tag=as045cd609
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=as5951033c'
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog
'[email protected]' Method: INVITE
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog
'[email protected]' Method: REGISTER
^@
^[[KWBPBXFG000304*CLI>
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