Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4.0
FreePBX = 2.11.0.2
In incoming routes I set the Alert-Info string to ‘alert-internal’. On extension 41712 I have these settings:
alert_internal_ring_text!: alert-internal alert_external_ring_text!: alert-external alert_group_ring_text!: alert-group alert_internal_ring_sound!: Ringer3 alert_external_ring_sound!: Ringer1 alert_group_ring_sound!: Ringer1
When I call in along this route I see this in asterisk -rvvvvvvvvvv:
-- Executing [ctclear@macro-dial-one:2] Return("DAHDI/1-1", "") in new stack -- Executing [s@macro-dial-one:30] Set("DAHDI/1-1", "D_OPTIONS=Ttr") in new stack -- Executing [s@macro-dial-one:31] ExecIf("DAHDI/1-1", "1?SIPAddHeader(Alert-Info: alert-internal)") in new stack -- Executing [s@macro-dial-one:32] ExecIf("DAHDI/1-1", "0?SIPAddHeader()") in new stack
However, I never see this header actually passed to the extension:
-- Executing [s@macro-dial-one:38] GotoIf("DAHDI/1-1", "1?godial") in new stack -- Goto (macro-dial-one,s,42) -- Executing [s@macro-dial-one:42] Dial("DAHDI/1-1", "SIP/41712&SIP/90012,20,Ttr") in new stack
And, on the extension, no such SIP header is received:
Received from udp:192.168.6.9:5060 at 2/7/2013 14:28:02:141 (566 bytes):OPTIONS sip:[email protected]:3072;line=o91i0f73 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK1bb78e95
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as3e57826d
To: sip:[email protected]:3072;line=o91i0f73
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.4.0)
Date: Tue, 02 Jul 2013 18:28:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
However, when I set Alert-Info to this:
http://nohost>\;info=alert-internal\;x-line-id=0/
As http://wiki.snom.com/Category:HowTo:Ringtone:Alert_Info indicates is necessary then I do see the SIP header on the phone sip trace log but not in the asterisk trace.
Received from udp:192.168.6.9:5060 at 2/7/2013 14:45:23:714 (937 bytes):
INVITE sip:[email protected]:3072;line=o91i0f73 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK38a01e4d
Max-Forwards: 70
From: “Private:” sip:[email protected];tag=as4d2c1800
To: sip:[email protected]:3072;line=o91i0f73
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Date: Tue, 02 Jul 2013 18:45:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Alert-Info: http://nohost>;info=alert-internal;x-line-id=0/
Content-Type: application/sdp
Content-Length: 278
This actually works (i.e the ring-tone chosen is the one that rings) but I still cannot see what is actually being send to the phone from the asterisk side. What asterisk command shows the exact string sent to the extension including the alert info setting?