Olle, Should I use defaultip and username when configuring gateways that do not register to *? Could this be why my gateways are not sending calls to the appropriate contexts?
Here is my sip.conf configuration for my gateway: ;----------------------- ; sip.conf ;----------------------- [general] context=default srvlookup=yes dtmfmode=inband qualify=yes nat=yes host=dynamic canreinvite=no pedantic=no disallow=all allow=ulaw allow=g729 allow=g723 allow=alaw [ 12.34.43.3] context=customer1 type=friend qualify=200 host= 12.34.43.3 canreinvite=no Thanks, Sum Ding Wong On 2/1/06, Olle E Johansson <[EMAIL PROTECTED]> wrote: > Chris A. Icide wrote: > > Ronald Wiplinger wrote: > > > > <snip> > > > >> > >>601, 602, 605, 606, 608, 609, 610, 615 and 616 are in sip.conf > >>621 and 626 are in Real-time sip_buddies > >> > >>621 and 626 changes username back from name to number (name) in the > >>database, and never shows it in "sip show peer" > >> > >>615 changed username "Ronald office" to 615, although no change in > >>sip.conf > >> > >>Did anybody else experienced that? > >> > >>*CLI> show version > >>Asterisk SVN-trunk-r8447M built by root @ vpbx on a x86_64 running > >>Linux on 2006-01-25 15:33:01 UTC > >> > > > > <snip> > > > > There is some code in asterisk which I'm not sure why it exists, that > > will set the username in memory to the user value in the SIP Contact > > header upon registration. While this isn't normally a big deal, if you > > are using realtime, when a SIP UA registers, some things are written > > back to the realtime database, username being one of them. > > > > I am not sure if this is a bug or not, as I don't understand the thought > > process behind allowing a sip ua to modify the username asterisk uses > > based on a sip header when it registers. > > > > I went into the code and removed the username as a field that got > > written back to the realtime db upon registration and it fixed my problem. > > > DO NOT USE USERNAME! > > That feild is really something used together with defaultIP when we have > *no* registration. > > * When we have a registration, we use whatever the phone tells us in the > Contact: header, which is a SIP requirement. That is why we change it to > reflect the known contact, provided by the phone. > > * We never change the device name of the phone, just the address we use > to communicate with the phone. > > * This name is not used for authentication by a phone, it is not the > username you configure the phone with. > > * In most cases, there is no need to use the "username=" option in > sip.conf for phones. > > I will soon change this setting to "defaultuser" to make it even more > obvious that this is not anything you need to set. The phone name is > whatever you have between the square brackets in sip.conf, both for > users and peers. > > I will check into the realtime code as well, to make sure we are > changing the proper field in the database. > > /Olle > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
