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.
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