Asterisk, SIP, отсутствие early media из-за повторного ответа «183 Session Progress» без SDP

Дамп пакетов, показывает, что сообщение о недоступности абонента действительно проигрывается провайдером, дебаг SIP-сообщений показывает, что через несколько секунд после ответа «183 Session Progress» приходит ещё один, но без SDP:

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.101:5060;branch=z9hG4bK27e2c144
Max-Forwards: 70
From: <sip:[email protected]>;tag=as0c82959c
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Mon, 18 Aug 2014 07:36:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 879613085 879613085 IN IP4 192.168.100.101
s=Asterisk
c=IN IP4 192.168.100.101
t=0 0
m=audio 39082 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/mts/89991112222

<--- SIP read from UDP:10.5.30.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.101:5060;branch=z9hG4bK27e2c144
From: <sip:[email protected]>;tag=as0c82959c
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE

<------------->
--- (6 headers 0 lines) ---

<--- SIP read from UDP:10.5.30.200:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.101:5060;branch=z9hG4bK27e2c144
From: <sip:[email protected]>;tag=as0c82959c
To: <sip:[email protected]>;tag=m0mbaknk-CC-27
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060;user=phone;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 190
Content-Type: application/sdp

v=0
o=- 3980427012 18526325 IN IP4 10.5.30.200
s=-
c=IN IP4 10.5.30.200
t=0 0
m=audio 42638 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 9 lines) ---
list_route: hop: <sip:[email protected]:5060;user=phone;transport=udp>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.5.30.200:42638
    -- SIP/mts-00000031 is making progress passing it to IAX2/voxlink-617

<--- SIP read from UDP:10.5.30.200:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.101:5060;branch=z9hG4bK27e2c144
From: <sip:[email protected]>;tag=as0c82959c
To: <sip:[email protected]>;tag=m0mbaknk-CC-27
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:[email protected]:5060;user=phone;transport=udp>
Content-Length: 0

IP-адреса и номера изменены: 192.168.100.101 — сервер Asterisk, 10.5.30.200 — сервер провайдера, 1111111111 — вызывающий номер, 89991112222 — вызываемый номер

Оказалось, Asterisk обрабатывает ответ «183 Session Progress» без SDP, как ответ «180 Ringing» из-за поведения АТС Alcatel:
(исходник Asterisk 11.11.0, channels/chan_sip.c, начиная со строки 22972)

...
        } else {
            /* Alcatel PBXs are known to send 183s with no SDP after sending
             * a 100 Trying response. We're just going to treat this sort of thing
             * the same as we would treat a 180 Ringing
             */
            if (!req->ignore && p->owner) {
                ast_queue_control(p->owner, AST_CONTROL_RINGING);
            }
        }
...

Меняем в этом отрывке AST_CONTROL_RINGING на AST_CONTROL_PROGRESS:

...
        } else {
            /* Alcatel PBXs are known to send 183s with no SDP after sending
             * a 100 Trying response. We're just going to treat this sort of thing
             * the same as we would treat a 180 Ringing
             */
            if (!req->ignore && p->owner) {
                ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
            }
        }
...

Пересобираем Asterisk, сообщение о недоступности номера в early media начинает нормально проигрываться звонящему.