OK, thanks. DNS is running on the sipXecs server.
I re-installed and used the fully qualified domain during the setup. The
auto generated DNS zone file is correct and all the tests pass.
However, I still have the initial problem of not being able to transfer
to an incoming call to an ext from the Auto Attendant.
Below is the contents of sipXproxy.log for the call. Some things that I
noticed are
* repeated warnings of "missing 'signature' param"
* DNS query for name '_sip._tls.voip.datatek-net.com', type = 33
(SRV): returned error ----->This SRV record is not in the zone file,
is this something needed by polycom phones?
* KERNEL:ERR ... doSendCore message send failed for queue
'SipTcpServer-3' - no room, ret = 9
sipXproxy.log:
"2012-03-21T14:59:18.826072Z":756:SIP:WARNING:voip.datatek-net.com:SipXProxyCseObserver-10:B77FAB90:SipXProxy:"SipXauthIdentity::decode
'<sip:[email protected]>' missing 'signature' param"
"2012-03-21T14:59:18.827981Z":757:SIP:WARNING:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipXauthIdentity::decode
'<sip:[email protected]>' missing 'signature' param"
"2012-03-21T14:59:18.829672Z":758:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send
outgoing call 1"
"2012-03-21T14:59:18.857719Z":759:SIP:WARNING:voip.datatek-net.com:SipSrvLookupThread-14:B6A74B90:SipXProxy:"DNS
query for name '_sip._tls.voip.datatek-net.com', type = 33 (SRV):
returned error"
"2012-03-21T14:59:18.867593Z":760:SIP:WARNING:voip.datatek-net.com:SipXProxyCseObserver-10:B77FAB90:SipXProxy:"SipXauthIdentity::decode
'<sip:[email protected]>' missing 'signature' param"
"2012-03-21T14:59:18.872363Z":761:SIP:WARNING:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"
"2012-03-21T14:59:18.873147Z":762:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send
outgoing call 1"
"2012-03-21T14:59:18.937284Z":763:SIP:ERR:voip.datatek-net.com:SipUserAgent-2:B6E78B90:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"
"2012-03-21T14:59:18.947827Z":764:SIP:WARNING:voip.datatek-net.com:SipXProxyCseObserver-10:B77FAB90:SipXProxy:"SipXauthIdentity::decode
'\"IVR\" <sip:[email protected]>' missing 'signature' param"
"2012-03-21T14:59:18.954528Z":765:SIP:WARNING:voip.datatek-net.com:SipXProxyCseObserver-10:B77FAB90:SipXProxy:"SipXauthIdentity::decode
'\"IVR\" <sip:[email protected]>' missing 'signature' param"
"2012-03-21T14:59:19.140082Z":766:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send
outgoing call 1"
"2012-03-21T14:59:30.718114Z":767:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send
outgoing call 1"
"2012-03-21T14:59:30.729993Z":768:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send
outgoing call 1"
"2012-03-21T14:59:46.579296Z":769:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send
outgoing call 1"
"2012-03-21T15:01:41.158371Z":770:KERNEL:ERR:voip.datatek-net.com:SipClientTcp-377:FFFFFFFF:SipXProxy:"OsMsgQShared::doSendCore
message send failed for queue 'SipTcpServer-3' - no room, ret = 9"
Any ideas?
Stiles
On 03/19/2012 04:40 PM, Douglas Hubler wrote:
Nettica.com should work.
If you only have one server, you don't need srv records, but dns
should be running on sipx server and it should be configured automatically
On Mar 19, 2012 3:53 PM, "Stiles Watson" <[email protected]
<mailto:[email protected]>> wrote:
Are there instructions for setting up SRV records with Godaddy? I
do not plan to ever ad servers, but if this is the preferred way.
If no one like Godaddy for this, are there DNS providers which
work well for this setup?
I tried a couple of months ago to get SRV recs set up, but I had a
problem with the _sip._tcp.rr record I asked about in my original
email. Godaddy is not setup to except it.
Stiles
On 03/19/2012 02:38 PM, Michael Picher wrote:
Changing the SIP domain is a bad thing to do...
If you want to use A-Records, you should start that way.
I'd suggest another reinstall and do it with A-Records....
Be aware if you use A-Records you can not just add servers to the
cluster. This assumes SRV records.
Mike
On Mon, Mar 19, 2012 at 2:31 PM, Stiles Watson
<[email protected] <mailto:[email protected]>> wrote:
This is a follow up question to a problem I submitted about 2
months
ago. Long story short, I reinstalled sipxecs 4.2 and made the
sipxecs
server the DHCP and DNS servers as well. This solved some of
my issues,
but not all. I still am unable to transfer incoming calls to any
extension. I can dial out from any ext, I can dial ext to
ext, I can
dial into the system and the Auto Attendant answers, but when
I dial an
ext I get "Please hold while I transfer your call" but the
ext never
rings. The CDR says the call was transfered and there is no
error reported.
All calls in and out of sipxecs are through a SIP Trunk via
Flowroute.
I'm using Polycom SoundPoint IP 335 phones. My firewall is a
Sonicwall
NSA, with SIP Transformations turned off and H.323 turned
off, and
consistent NAT enabled. RTP ports 30000-31000 UDP, SIP port
5060 UDP &
TCP, SIP port 5080 UDP all open and NAT rules set up to send
traffic on
these ports to the sipxecs server. I have to support remote
phones.
My external DNS records are through godaddy and the only
record for
sipxecs is an A record. I've followed the instructions at
http://wiki.sipfoundry.org/display/sipXecs/DNS+Concepts+for+sipXecs
for
setting up A records and have changed the domain name to the
fully
qualified domain under System-->Domain. I clicked Apply,
restarted the
services and then rebooted. I also sent new profiles to the
phones. This
also required me to change the internal DNS zone file to use
the fully
qualified domain. After doing this I restarted the named service.
DNS advisor runs successfully, but the DNS test under
Diagnostics-->Configuration Tests returns a warning that SRV
records for
the unqualified domain can not be found:
Starting DNS servers test.
DNS testing of name server: voip.datatek-net.com
<http://voip.datatek-net.com>
NAPTR Lookup for datatek-net.com <http://datatek-net.com>
domain:
No NAPTR records found.
SRV Lookup for Target: _sip._udp.datatek-net.com
<http://udp.datatek-net.com>
No SRV records found.
SRV Lookup for Target: _sip._tcp.datatek-net.com
<http://tcp.datatek-net.com>
No SRV records found.
SRV Lookup for Target: _sips._tcp.datatek-net.com
<http://tcp.datatek-net.com>
No SRV records found.
Why is the test looking for SRV records for datatek-net.com
<http://datatek-net.com> and not
voip.datatek-net.com <http://voip.datatek-net.com> (which
exist and which I had to add to make the DNS
advisor happy)?
There is one record in the internal DNS config that I do not
understand:
; SRV record for service SIP TCP rr.voip.datatek-net.com
<http://rr.voip.datatek-net.com>
; priority: 1 weight: 0 port: 5070 server:
voip.datatek-net.com <http://voip.datatek-net.com>
;
_sip._tcp.rr.voip.datatek-net.com
<http://tcp.rr.voip.datatek-net.com>. IN SRV 1 0 5070
voip.datatek-net.com <http://voip.datatek-net.com>.
Why is port 5070 being used and for what?
Thanks for your help...
Stiles
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