1я проблема решилась - вкладка digit map, default call route поменял с Auto(first voip) на Voip
2я проблема по прежнему актуальна, причем если звонок со стороны шлюза и отбивается астериском, то все нормально. если же звонок со стороны астериска - то шлюз не видит bye
при посылке bye астериском на dlink'e нет никаких логов
трейс с астериска :
Код:
Reliably Transmitting (no NAT) to 10.5.73.20:5060:
INVITE sip:89021772220@10.5.73.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.73.1:5060;branch=z9hG4bK0bd5bfa8
Max-Forwards: 70
From: "TEST" <sip:3603@10.5.73.1>;tag=as5a6b2235
To: <sip:89XXXXXXXXX@10.5.73.20:5060>
Contact: <sip:3603@10.5.73.1:5060>
Call-ID: 56df380d348e6db953c417a2550acdbc@10.5.73.1:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Sat, 09 Jun 2012 03:22:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 325
v=0
o=root 1599351030 1599351030 IN IP4 10.5.73.1
s=Asterisk PBX 1.8.11.0
c=IN IP4 10.5.73.1
t=0 0
m=audio 15930 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:10.5.73.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.5.73.1:5060;branch=z9hG4bK0bd5bfa8
From: "TEST" <sip:3603@10.5.73.1>;tag=as5a6b2235
To: <sip:89XXXXXXXXX@10.5.73.20:5060>
Call-ID: 56df380d348e6db953c417a2550acdbc@10.5.73.1:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.5.73.20:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 10.5.73.1:5060;branch=z9hG4bK0bd5bfa8
From: "TEST" <sip:3603@10.5.73.1>;tag=as5a6b2235
To: <sip:89XXXXXXXX@10.5.73.20:5060>;tag=62b2f524-686034
Call-ID: 56df380d348e6db953c417a2550acdbc@10.5.73.1:5060
CSeq: 102 INVITE
Contact: <sip:4801@10.5.73.20:5060>
User-Agent: dlink 12-3868-8185-1.2.1.1835-SA7O4
Content-Type: application/sdp
Content-Length: 210
v=0
o=4802 1793229230 1793229230 IN IP4 10.5.73.20
s=Session SDP
c=IN IP4 10.5.73.20
t=0 0
m=audio 9004 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
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 - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.5.73.20:9004
list_route: hop: <sip:4801@10.5.73.20:5060>
set_destination: Parsing <sip:4801@10.5.73.20:5060> for address/port to send to
set_destination: set destination to 10.5.73.20:5060
Transmitting (no NAT) to 10.5.73.20:5060:
ACK sip:4801@10.5.73.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.73.1:5060;branch=z9hG4bK68975ef8
Max-Forwards: 70
From: "TEST" <sip:3603@10.5.73.1>;tag=as5a6b2235
To: <sip:89XXXXXXXXX@10.5.73.20:5060>;tag=62b2f524-686034
Contact: <sip:3603@10.5.73.1:5060>
Call-ID: 56df380d348e6db953c417a2550acdbc@10.5.73.1:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.11.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '56df380d348e6db953c417a2550acdbc@10.5.73.1:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:4801@10.5.73.20:5060> for address/port to send to
set_destination: set destination to 10.5.73.20:5060
Reliably Transmitting (no NAT) to 10.5.73.20:5060:
BYE sip:4801@10.5.73.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.73.1:5060;branch=z9hG4bK6ecf48c0
Max-Fo"TEST" <sip:3603@10.5.73.1>;tag=as5a6b2235
To: <sip:89XXXXXXXXX@10.5.73.20:5060>;tag=62b2f524-686034
Call-ID: 56df380d348e6db953c417a2550acdbc@10.5.73.1:5060
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:10.5.73.20:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.5.73.1:5060;branch=z9hG4bK6ecf48c0
From: "TEST" <sip:3603@10.5.73.1>;tag=as5a6b2235
To: <sip:89XXXXXXXXX@10.5.73.20:5060>;tag=62b2f524-686034
Call-ID: 56df380d348e6db953c417a2550acdbc@10.5.73.1:5060
CSeq: 103 BYE
User-Agent: dlink 12-3868-8185-1.2.1.1835-SA7O4
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '56df380d348e6db953c417a2550acdbc@10.5.73.1:5060' Method: INVITE
Elastix-2*CLI> sip set debug off
SIP Debugging Disabled
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