I have read the WIKI and tried several times to get the Ringtones to modify on this phone based on the call-type hitting the phone… it seems not to take my alert info data… here is how I have my digium phones.conf file set up.
I am using asterisk 1.8.11-cert5 and the latest DPMA as well as the latest phone firmware.
below being the idea that I can have a single or double ring using the built in ring-tones on the phones. these are my alerts.
I did a reconfigure on the phone after adding these settings, and even power cycled it, as well as did a CLI reload of the phones module and power cycled the phone again afterwords…
[alert-office-single]
type=alert
alert_info=office-single
ring_type=normal
ringtone=Office
[alert-office-double]
type=alert
alert_info=office-double
ring_type=normal
ringtone=Office2
I set up a sip header add when a call comes into the phone to update to call the alert. the way I understand it is I am supposed to in my alert info header send the value of alert_info= depending on what I want…
below is the CLI verbose of me placing a call to the phone… i can see the addheader being set… but whether I set it to office-single or office-double the phone rings the same way and doesnt change…
-- Executing [s@macro-ringphone:71] Set("SIP/6009-0000004b", "CONNECTEDLINE(all)=6005 <DIGIUM D50>") in new stack
-- Executing [s@macro-ringphone:72] SIPAddHeader("SIP/6009-0000004b", ""Alert-Info: <office-double>"") in new stack
-- Executing [s@macro-ringphone:73] Dial("SIP/6009-0000004b", "SIP/6005,90,I") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6005
-- Connected line update to SIP/6009-0000004b prevented.
-- SIP/6005-0000004c is ringing
and below the SIP debug shows the header being sent to the phone…
== Using SIP RTP CoS mark 5
Audio is at 15654
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.10.243:5060:
INVITE sip:6005@192.168.10.243:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.10.5:5060;branch=z9hG4bK5015bf89
Max-Forwards: 70
From: "Basement" <sip:6009@192.168.10.5>;tag=as36181fe2
To: <sip:6005@192.168.10.243:5060;ob>
Contact: <sip:6009@192.168.10.5:5060>
Call-ID: 60fed861128d3efe4365bfc46ea86bbf@192.168.10.5:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11-cert5
Date: Sun, 19 Aug 2012 17:00:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Alert-Info: <office-double>
Content-Type: application/sdp
Content-Length: 266
any ideas?
-Christopher