Yes, under Account Settings-->Inbound Settings, I set Device Type to "IP
PBX Server, Asterisk or Softswitch". Is this the only place to set it? I
was assuming this was an account level setting and not a DID setting.
Stiles
On 03/21/2012 05:19 PM, Michael Picher wrote:
in the voip.ms <http://voip.ms> did you change the account to be
'Asterisk or SIP PBX' (i'm paraphrasing).
mike
On Wed, Mar 21, 2012 at 5:04 PM, Stiles Watson <[email protected]
<mailto:[email protected]>> wrote:
We'll I signed up for voip.ms <http://voip.ms> and set up the
gateway for it and now all calls work in and out of sipxecs... but
now resume from hold and canceling a transfer result in losing the
call.
Again, I'm using Polycom SoundPoint IP 335 phones.
While on an active call, I can receive the call and put it on
hold, but there is no way to get the call back. If press hold
again, or the soft button to resume, audio in both directs is
lost. Same goes for pressing the soft button to transfer and then
cancel - no audio.
I can transfer the call successfully, it is only canceling the
transfer which does not work.
Also, many of the calls through voip.ms <http://voip.ms> are
'active' according to the CDR. I'm guessing these are the lost
trnasfers and holds? Some of them have been active for over 30 min
after I end the call.
Stiles
On 03/21/2012 12:29 PM, Stiles Watson wrote:
While searching the old messages on this list to see if I can get
any insight on this issue, I found this 2010 note from Tony
Graziano in a response to a different question:
"It means they are not acking the call. I suspect this is because
sipxbridge may not be involved in the call, and only sipxproxy
is, which would be problematic for a lot of call scenarios (like
transfers)."
This seems related to my issue. I see log entries in
sipXproxy.log for the failed transfers, but I do not see anything
in sipxbridge.log. Mostly I just see keep-alive records for
flowroute (US and CANADA) and callwithus (for international calls
only). It could be I do not understand what to look for...
Anyway, how would sipxbridge "not be involved in the call" and
how do I check to see if it is or not and how do I fix it?
At this point the only thing that does not work are incoming
calls answered by the AA can not transfer to an ext. The CDR says
the call was transfered, but it is. Calling out works, and ext to
ext works.
Stiles
On 03/21/2012 11:18 AM, Stiles Watson wrote:
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
<http://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]>
<mailto: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]>
<mailto: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
<http://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]>
<mailto: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]>
<mailto: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]>
<mailto: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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eZuce, Inc.
300 Brickstone Square
Suite 201
Andover, MA. 01810
O.978-296-1005 X2015 <tel:978-296-1005%20X2015>
M.207-956-0262 <tel:207-956-0262>
@mpicher <http://twitter.com/mpicher>
www.ezuce.com <http://www.ezuce.com>
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eZuce, Inc.
300 Brickstone Square
Suite 201
Andover, MA. 01810
O.978-296-1005 X2015
M.207-956-0262
@mpicher <http://twitter.com/mpicher>
www.ezuce.com <http://www.ezuce.com>
------------------------------------------------------------------------------------------------------------
There are 10 kinds of people in the world, those who understand binary
and those who don't.
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