I am getting a little confused here Tony, but I will reiterate what I see. I think I am seeing what you are seeing.
First time into a virgin devices groups / Aastra (55i say). Click into server settings then click on "show advanced settings" The proxy port, registrar port and outbound proxy port are all 0. This is what we want (for now). I have entered a FQDN for the outbound proxy as all my Aastras seem not to handle SRV DNS records. Now into my devices list (based on my devices group) and onto one of my Aastra phone lines. Click onto server and "show advanced settings" proxy and registrar are automatically filled in with the default domain name. In my case datamerge.local Outbound proxy is set to sipxecs.datamerge.local as inherited from my devices group. proxy port and registrar port are set to 5060 which is NOT what I want. Outbound proxy port is still 0 I have to manually set my proxy port and registrar port to 0 manually for each handset. If the phone has its proxy port and registrar port set to 5060 it presents the URI (e.g. for extension 204) as REGISTER sip:datamerge.local:5060, INVITE sip:mailto:[email protected]:5060 and SUBSCRIBE sip:mailto:[email protected]:5060. Registrations and INVITEs handle this fine, but SUBSCRIPTIONS fail with a 401 auth error. When the phone has its proxy port and registrar port set to 0 it presents the URI as REGISTER sip:datamerge.local, INVITE sip:mailto:[email protected] and SUBSCRIBE sip:mailto:[email protected]. This works for all functions. In all the above I set the outbound proxy on the Aastras to give the phone a FQDN (sipxecs.datamerge.local) to talk to as it seems to have difficulty with SRV lookups. I think the bug in the provisioning is that the default values in the devices groups screens are populated with 0, instead of being NULL like other fields. These are treated as NULLS by the template logic, where 0 is a valid value for the Aastras. The fields in the group template should be NULL instead of 0. When 0 is put into them the phone configs should follow with 0. Alternatively, the default of the phone settings should simply be 0 to follow the default group template. Hope this makes it clearer. Tony Graziano wrote on Tue, 07 August 2012 21:09 > but this also means the template was modified by the > user or the defaults > were changed somehow, because they default to 5060, > which does work. So > this issue is if "it is set to 0" it does not behave as > expected. > > While I agree "ideally" it would do a DNS lookup if set > to port 5060, but > would also need to "send" port 5060 if the DNS sends > this back. In as much > as I want to think this is a sipx thing, we'd really > need to understand the > logic and whether that is specifically supported by > Aastra to perform this > function. Why I say Aastra (dont take it as a bad thing > to say) is because > RIGHT NOW the hostname does not get set in the proxy or > registrar fields > for the line, just the sip domain. For subscriptions and > registrations it > sends/auths successfully as "user@sipdomain:port". > > So that is two things: will it send port 5060 if doing a > DNS lookup and > assumes if it finds port 5060, does it use this. That I > think is a UA > question. When it does that, does it also use the > sipdomain or the hostname? > > Ideally the lookup function would only be relevant to > the port number and > not change the proxy/registrar to hostname (IMO). > > What is the use case to change the port from 5060 to "0" > though? > > > > On Tue, Aug 7, 2012 at 8:04 AM, Michael Picher > <[email protected]> wrote: > > > The problem is specifically with Aastra phones that > > if port is set to 0 > > then it will use SRV, is port is set it will use > > A-Record resolution. The > > phone config template is putting in 5060 and this > > should be user definable > > and should default to 0. > > > > Mike > > > > On Tue, Aug 7, 2012 at 7:22 AM, Tony Graziano < > > [email protected]> wrote: > > > >> > >> > >> On Tue, Aug 7, 2012 at 7:13 AM, Mark Dutton > <[email protected]>wrote: > >> > >>> > >>> > >>> I did say exactly what I did to make it work in an > earlier > >>> post. The problem was that the SipX provisioning > software > >>> was not carrying the port = 0 variable in the > device group > >>> server settings for registrar and proxy. Even > though the > >>> device group profile has 0, the device profile > still puts in > >>> 5060 as a default. You have to go to each device > >>> individually and override the setting with 0. This > then > >>> causes the handset to use the default port of 5060, > but not > >>> to specify it in the URI. This then made the MWI > server > >>> happy to auth it. > >>> > >> > >> In normal instances, sipxecs tries to be "SRV > aware". Using port "0" > >> means to look up the information (automatic) and DNS > will find the correct > >> port and transport. > >> > >> " You have to go to each device individually and > override the setting > >> with 0. " Is this the case on 4.4 or 4.6 or both? > >> > >> I would help if you supplied the value for whatever > versions that you had > >> to manually change in the handset: > >> > >> i.e. sip line1 proxy port: 0 to sip line1 proxy > port: 5060 > >> > >> to get it to work. > >> > >> Because on my systems (4.4 and 4.6), it says this in > the line settings. > >> > >> > >> (i.e Phone, Line, Server Settings) > >> > >> sip line1 registrar port: 5060 > >> sip line1 proxy port: 5060 > >> > >> So I do not know what is different on yours. On mine > my DNS records are > >> fully populated, but both generated configs say > "5060" by default, > >> specifically, but I don't have an Aastra to test > with. The only thing that > >> has "0" in it is the outbound proxy port. I am > trying to understand why > >> yours defaults to "0" and mine does not. For what it > is worth, the > >> timezones are not fully populated to be of good > enough production use for > >> certain parts of the world either so it needs a > maintainer. > >> > >> > >> > >> > >> > >> > >> > >> > >> > >>> And I can tell you with all certainty that Zultys > does not > >>> use a special firmware. I have been a Zultys beta > tester for > >>> 5 years working with the dev guys. Aastra has a > provisioning > >>> system where it goes to the Internet on first boot > (after > >>> factory default) and looks up the mac address a > database > >>> maintained by Aastra. Aastra then sends back > certain > >>> identity information, such as the correct splash > screen > >>> bitmap and agent string, etc. > >>> > >>> The actual firmware is direct from Aastra and is > unmodified > >>> (in the Zultys case). > >>> > >>> What got my back up was that in my first post I > asked what > >>> information to gather to send to the SipX forum and > instead > >>> I was told to send a SIP log to Aastra. Believe me > they > >>> would have absolutely no interest in even > replying. > >>> > >> > >> I never suggested that. I did suggest a siptrace > from sipx so the call > >> flow could be seen using sipviewer. > >> > >>> > >>> I am new to SipX, but not to IP tel. I am not sure > which > >>> logs give me what sort of information (apart from > >>> sipXproxy.log). I have been using sipx-trace and > sipviewer > >>> (when necessary) to do my investigations to date. > Where I > >>> came unstuck with this was that I did not know why > I was > >>> getting an auth error. > >>> > >>> As it turned out, it was because the phone was > subscribing > >>> with the port in the URI. From Joegen's post (the > RFC > >>> excerpt), I could see that the port designator is a > key part > >>> of a URI match and this is what SipX didn't like. > However, > >>> if SipX was being strict, it should not have > allowed the > >>> REGISTER, or INVITE methods either as these too > were > >>> appending the port to the URI. > >>> > >>> One has to be pragmatic with SIP. There are so > many > >>> "viewpoints" on what is legal. When it comes to > Aastra, a > >>> large, isolationist company, or SipX, an open > community, it > >>> is going to be the latter that is more likely to > accomodate > >>> change than the former. Just the way of the world. > >>> -- > >>> Regards > >>> > >>> Mark Dutton > >>> > >>> _______________________________________________ > >>> sipx-users mailing list > >>> [email protected] > >>> List Archive: > http://list.sipfoundry.org/archive/sipx-users/ > >>> > >> > >> > >> > >> -- > >> ~~~~~~~~~~~~~~~~~~ > >> Tony Graziano, Manager > >> Telephone: 434.984.8430 > >> sip: [email protected] > >> Fax: 434.465.6833 > >> ~~~~~~~~~~~~~~~~~~ > >> Linked-In Profile: > >> > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > >> Ask about our Internet Fax services! > >> ~~~~~~~~~~~~~~~~~~ > >> > >> Using or developing for sipXecs from SIPFoundry? Ask > me about sipX-CoLab > >> 2013! > >> > <http://sipxcolab2013.eventbrite.com/?discount=tony2013> > >> > >> > >> LAN/Telephony/Security and Control Systems > Helpdesk: > >> Telephone: 434.984.8426 > >> sip: > [email protected].**net<[email protected]> > >> > >> Helpdesk Customers: > http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> > >> Blog: http://blog.myitdepartment.net > >> > >> _______________________________________________ > >> 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> > > linkedin < > > http://www.linkedin.com/profile/view?id=35504760&trk=tab > > _pro> > > 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/ > > > > > > -- > ~~~~~~~~~~~~~~~~~~ > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > ~~~~~~~~~~~~~~~~~~ > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > ~~~~~~~~~~~~~~~~~~ > > Using or developing for sipXecs from SIPFoundry? Ask me > about sipX-CoLab > 2013! > > <http://sipxcolab2013.eventbrite.com/?discount=tony2013> > > -- > 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 > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: > http://list.sipfoundry.org/archive/sipx-users/ -- Regards Mark Dutton _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
