I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. "sip show subscriptions" shows nothing.
I just started playing with it this morning however...still playing around w/ the configs. One odd thing, I keep seeing some weirdness: [Mar 5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default (domain x.x.x.x) And also Looking for 103 Yet I have no idea where those values are coming from! I am running 1.6.7. Here is a snippet of the phone config from one of the phones: <reg reg.1.displayName="Line 1" reg.1.address="station2_line1" reg.1.label="Line 1" reg.1.type="shared" reg.1.thirdPartyName="2404366402-2" reg.1.auth.userId="2404366402-2" reg.1.auth.password="1234" reg.1.server.1.address="" reg.1.server.1.port="" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="60" reg.1.server.1.register="1 I noticed that I had to set the reg.x.address field to the stationX_lineX value or the phone wouldn't fill in the "icon" image...but it would accept cals. Still not completely clear but I am making progress! Bill -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Friday, March 02, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Lacy Moore - Aspendora wrote: > Russell, I don't have any specifics at this time. I need to dig a > little further. I'm thinking the autocontext is what is giving me > fits. I can receive calls and place calls, but the hint status is not > working. It currently registers as a hint showing not in use. It > does not show in use. If you aren't seeing any lights change on the phones when calls are going on, check "sip show subscriptions" at the CLI. If the phones have not properly subscribed to the right extensions, you won't see anything. > I ended up using some of the config from the bottom of the sla.txt > file. The sample file may be missing the template section. The > sample config does not match the config in the sla.txt. I couldn't > get the sample config to work at all. Again, hopefully over the > weekend I'll be able to get more information. You are correct. The sample configuration is missing the template. I will add it now. However, I just made the tarballs for 1.4.1, so this config fix didn't make it in. > Using the config in the sample file, the hint status was working. I > could see the line ringing, but I could not answer the lines or place > calls. Using the config from the sla.txt file, I could place calls > and receive calls, but the hints never showed any activity, just > always not in use. As I noted earlier, check your "sip show subscriptions" to make sure the phones are subscribed to the right thing. Another helpful thing that you can use for debugging is to look at the output of "sla show stations". You can see the state of each line appearance on each station. This should correspond with what you see on the phone ... unless there is a problem, of course. > If possible, could you provide the config that you've used for > testing? I'm testing using Polycom phones to try to keep things > simple. I'm assuming you are using a Polycom. I have been testing with a variety of different phones. I have not tested all of the Polycom models, yet. This is one of the things we're going to have to work through. I would like to document issues with specific phones in sla.txt as we come across them. The configuration I'm using for testing looks just like the stuff in configs/sla.conf.sample. Essentially, it is: [line1] type=trunk device=Zap/3 autocontext=line1 [line2] type=trunk device=Zap/4 autocontext=line2 [station](!) type=station autocontext=sla_stations trunk=line1 trunk=line2 [station1] (station) device=SIP/station1 [station2](station) device=SIP/station2 [station3](station) device=SIP/station3 Thanks for providing some feedback on this. You are the first one to say anything about it. :) I am very eager to get everything working well so that everyone is happy. Just please be patient as I work through the initial flood of reports since it is just now getting out in the field. -- Russell Bryant Software Engineer Digium, Inc. _______________________________________________ --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
