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<?xml version="1.0" ?>
<config>
<ringtones>
<alerts>
<alert alert_info="normal" ringtone_id="Digium" ring_type="normal" />
<alert alert_info="ring-answer" ringtone_id="Digium" ring_type="ring-answer" />
<alert alert_info="intercom" ringtone_id="" ring_type="answer" />
<alert alert_info="visual" ringtone_id="" ring_type="visual" />
</alerts>
</ringtones>
</config>
exten => 100,1,Dial(PJSIP/alice,,b(pagehandler^addheader^1))
[pagehandler]
exten => addheader,1,Set(PJSIP_HEADER(add,Alert-Info)=<intercom>)
[alert-google]
type=alert
alert_info=ai-Google
ring_type=normal
ringtone=RotaryPhone
[d40office]
type=phone
network=mynetwork
mac=000fd3069e2e
line=10010
line=10050
full_name=My Name
timezone=America/Louisville
ntp_resync=86400
parking_exten=70
parking_transfer_type=blind
web_ui_enabled=true
record_own_calls=yes
send_to_vm=yes
active_locale=en_US
alert=alert-wakeup
alert=alert-google
blf_unused_linekeys=no
same => n,SipAddHeader("Alert-Info: ai-Google")
option boot-server code 66 = string;
...
option boot-server "http://192.168.0.7:8080/phoneprov/";
I'm not sure why this was necessary, since I don't have DPMA.
However, I now have some minor problems. I used the example in the XML Configuration 1.4.0 document as the basis for my configuration file. I cut and pasted as needed into my mac_addr.cfg file, updating elements as needed. BTW, there is an error on line 56, in <setting id="sip_qos". The value of 1 is not completely enclosed in double quotes. I found this with xmllint. My resulting XML configuration file also passes muster with xmllint.
Now when the phone restarts and eventually registers with Asterisk, it does not display the line labels for the line buttons in the upper right of the display. The icon of a person does not appear after successful registration, also. The label areas are completely blank. I coded values for line_label in <account> in <accounts>. The entered value shows up on the phone's web interface, but not on the actual phone. I know the phone registers. It displays a red blinking MWI light and the correct number of outstanding e-mails is displayed. Asterisk also shows it as registered. It also shows both lines as peers. Any ideas?
Also, when I call the phone from another SIP phone (a softphone on a laptop next to the Digium phone), in addition to getting the desired ringtone, the phone answers on the first ring thru the speaker. I never get to lift up the hand set.
Once the phone has a config on it, a firmware update or downgrade isn't going to override the config source that the phone's using. You really wouldn't want that to happen in practice. As you'd already loaded some config on your phone Option 66 was ignored.
The value in the example is 3. What's the URL of the page with the example you had a problem with?
What does your entire account element look like?
<!-- Accounts Element - - Define account info for registration to SIP server -->
<accounts>
<!-- Account for Line 1 -->
<account index="0" status="1" register="1" account_id="D40Line1" username="D40Line1"
authname="D40Line1" password="xxxx" passcode="xxxx"
line_label="Line 1" caller_id="D40 Line 1 <401>"
dial_plan="" conflict="replace" >
<host_primary server="192.168.0.7" port="5060" transport="udp" reregister="300"
retry="25" num_retries="5" />
<permission id="record_own_calls" value="0" /> <!-- Only used on DPMA -->
</account>
<!-- Account for Line 2 -->
<account index="1" status="1" register="1" account_id="D40Line2" username="D40Line2"
authname="D40Line2" password="xxxx" passcode="xxxx"
line_label="Line 2" caller_id="D40 Line 2 <402>" >
<host_primary server="192.168.0.7" port="5060" transport="udp" reregister="300"
retry="25" num_retries="5" />
<permission id="record_own_calls" value="0" /> <!-- Only used on DPMA -->
</account>
</accounts>
That's because you've set a ring_type of "ring-answer"
If you want a different behavior, you'll need to use a different ring_type
charleyb wrote:Now, is there anything else anywhere in the XML configuration file that can indicate or specify the number of lines to be configured in a phone?
<?xml version="1.0" ?>
<config>
<accounts>
<account index="0" status="1" register="1" account_id="100" username="100" authname="100" password="100" passcode="100" line_label="100" caller_id="100" dial_plan="" visual_voicemail="0" voicemail="sip:800@pbx.example.com">
<host_primary server="pbx.example.com" port="5060" transport="udp" />
</account>
</accounts>
</config>
I take it that only the <accounts> element and its subelements affect the number of lines (or places for lines) shown on the phone display. Is this a correct assumption?
To be sure I have a level playing field, I take it that resetting the phone to factory defaults clears as much from the phone (from previous configurations) as possible. The method I have been using is to go to the phone display, select "more", then "menu", then "Advanced", then "Reset to factory default". Is the correct way? There's not a magic button somewhere on the phone that I have to stick an unbent paperclip wire into, to reset the phone?
You mention that accounts are loaded in index order. I have a D40 two-line phone. When I specify an account with an index of 0 and another with an index of 9 (beyond the range of any Digium phone), the second account acts like a second line, as if I coded index of 1. It accepts calls directed to that account's username and shows as that user when the phone is used to create an outgoing call to another SIP phone (a softphone on a laptop). I take it that default behavior for the index value is to assign it to the last line on the phone?
What would the expected behavior be if I removed all account information, including the <accounts> element?
<setting id="desi_strip_enable" value="1" >
<setting id="desi_strip_enable" value="0" > <-- D40 does not have sidecar -->
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