http://blog.myitdepartment.net/?p=191 On Mar 21, 2012 5:29 PM, "Michael Picher" <[email protected]> wrote:
> in the 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]>wrote: > >> We'll I signed up for 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 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', 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]> <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]> <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]> <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]> <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]> <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]> 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]>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 >>>> NAPTR Lookup for datatek-net.com domain: >>>> No NAPTR records found. >>>> >>>> SRV Lookup for Target: _sip._udp.datatek-net.com >>>> No SRV records found. >>>> >>>> SRV Lookup for Target: _sip._tcp.datatek-net.com >>>> No SRV records found. >>>> >>>> SRV Lookup for Target: _sips._tcp.datatek-net.com >>>> No SRV records found. >>>> >>>> Why is the test looking for SRV records for datatek-net.com and not >>>> 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 >>>> ; priority: 1 weight: 0 port: 5070 server: voip.datatek-net.com >>>> ; >>>> _sip._tcp.rr.voip.datatek-net.com. IN SRV 1 0 5070 >>>> voip.datatek-net.com. >>>> >>>> Why is port 5070 being used and for what? >>>> >>>> Thanks for your help... >>>> >>>> Stiles >>>> >>>> _______________________________________________ >>>> sipx-users mailing list >>>> [email protected] >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>> >>> >>> >>> -- >>> Michael Picher, Director of Technical Services >>> eZuce, Inc. >>> >>> 300 Brickstone Square >>> >>> Suite 201 >>> >>> Andover, MA. 01810 >>> O.978-296-1005 X2015 <978-296-1005%20X2015> >>> M.207-956-0262 >>> @mpicher <http://twitter.com/mpicher> >>> www.ezuce.com >>> >>> >>> ------------------------------------------------------------------------------------------------------------ >>> There are 10 kinds of people in the world, those who understand binary >>> and those who don't. >>> >>> >>> >>> _______________________________________________ >>> sipx-users mailing [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> _______________________________________________ >> sipx-users mailing [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> >> _______________________________________________ >> sipx-users mailing [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> >> _______________________________________________ >> sipx-users mailing [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > Michael Picher, Director of Technical Services > 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 > > > ------------------------------------------------------------------------------------------------------------ > There are 10 kinds of people in the world, those who understand binary and > those who don't. > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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