Добрый день,
Есть следующий путь XLite=>Mera=>Asterisk=>GW При звонке c XLite сразу происходит отбой вызова, затем Asterisk звонит через шлюз по набранному номеру, но Xlite уже отключен и звонок происходит как бы сам в себе в Астериске. Если вызов принять - то тишина. Посоветуйте пожалуйста что можно с этим сделать?
SIP.conf
[810]
disallow=all
register=> 711:***:10007@AsteriskIntIP/711
type=peer
fromuser=711
authname=711
defaultuser=711
username=711
secret=ss9045522004
fromdomain=*meraIP*
host=*meraIP*
port=5060
canreinvite=no
callerid=711 <711>
nat=yes
dtmfmode=rfc2833
insecure=port,invite
reinvite=no
;context=hg
allow=alaw,ulaw,g729
context=from-internal
[general]
vmexten=*97
faxdetect=yes
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.10.1(1.8.19.1)
disallow=all
allow=ulaw
allow=alaw
allow=g729
callevents=yes
jbenable=no
defaultexpiry=120
maxexpiry=3600
minexpiry=60
allowguest=no
srvlookup=no
registerattempts=0
registertimeout=20
notifyhold=yes
rtptimeout=30
g726nonstandard=no
t38pt_udptl=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
nat=yes
externip=AsteriskExtIP
localnet=AsteriskIntIP/255.255.255.0
extension.conf
[from-internal]
exten=>_81189XXXXXXXXX,1,Dial(SIP/${EXTEN:0:3}/*${EXTEN:2:1}${EXTEN:3})
exten=>_81189XXXXXXXXX,n,Hangup()
[from-sip-internal]
exten=>_81189XXXXXXXXX,1,Dial(SIP/${EXTEN:0:3}/*${EXTEN:2:1}${EXTEN:3})
exten=>_81189XXXXXXXXX,n,Hangup()
===========sip set debug peer 711
[2013-03-25 02:35:52] VERBOSE[4731] asterisk.c: -- Remote UNIX connection
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c:
<--- SIP read from UDP:MeraExtIP:5060 --->
INVITE sip:81189*********@AsteriskExtIP:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Contact: <sip:10007@MeraExtIP;user=phone>
Max-Forwards: 10
User-Agent: MERA MSIP v.1.0.2
Cisco-Guid: 2284985600-3464449808-2147483696-1216657376
Content-Type: application/sdp
Content-Length: 266
v=0
o=- 1364164558 1364164558 IN IP4 MeraExtIP
s=-
c=IN IP4 MeraExtIP
t=0 0
m=audio 28632 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: --- (12 headers 12 lines) ---
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Sending to MeraExtIP:5060 (NAT)
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Using INVITE request as basis request - 40a42500ce7f4f10800000304884b7e0@mera
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found peer '810' for '10007' from MeraExtIP:5060
[2013-03-25 02:35:59] VERBOSE[4744] netsock2.c: == Using SIP RTP TOS bits 184
[2013-03-25 02:35:59] VERBOSE[4744] netsock2.c: == Using SIP RTP CoS mark 5
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found RTP audio format 18
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found RTP audio format 8
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found RTP audio format 0
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found RTP audio format 4
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found RTP audio format 101
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found audio description format G729 for ID 18
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found audio description format PCMA for ID 8
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found audio description format PCMU for ID 0
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found audio description format G723 for ID 4
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Found audio description format telephone-event for ID 101
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Peer audio RTP is at port MeraExtIP:28632
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: Looking for 81189********* in from-internal (domain AsteriskExtIP)
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: list_route: hop: <sip:10007@MeraExtIP;user=phone>
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c:
<--- Transmitting (NAT) to MeraExtIP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Length: 0
<------------>
[2013-03-25 02:35:59] VERBOSE[4854] pbx.c: -- Executing [81189*********@from-internal:1] Dial("SIP/810-0000000a", "SIP/811/*189*********") in new stack
[2013-03-25 02:35:59] VERBOSE[4854] netsock2.c: == Using SIP RTP TOS bits 184
[2013-03-25 02:35:59] VERBOSE[4854] netsock2.c: == Using SIP RTP CoS mark 5
[2013-03-25 02:35:59] VERBOSE[4854] app_dial.c: -- Called SIP/811/*189*********
[2013-03-25 02:35:59] VERBOSE[4854] app_dial.c: -- SIP/811-0000000b is ringing
[2013-03-25 02:35:59] VERBOSE[4854] chan_sip.c:
<--- Transmitting (NAT) to MeraExtIP:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Length: 0
<------------>
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c:
<--- SIP read from UDP:MeraExtIP:5060 --->
CANCEL sip:81189*********@AsteriskExtIP:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP MeraExtIP:5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 CANCEL
Max-Forwards: 10
User-Agent: MERA MSIP v.1.0.2
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0
<------------->
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: --- (10 headers 0 lines) ---
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c:
<--- Transmitting (NAT) to MeraExtIP:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP MeraExtIP:5060;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as35687798
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 CANCEL
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c:
<--- SIP read from UDP:MeraExtIP:5060 --->
ACK sip:81189*********@AsteriskExtIP:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP MeraExtIP:5060;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as35687798
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 ACK
Max-Forwards: 10
User-Agent: MERA MSIP v.1.0.2
Content-Length: 0
<------------->
[2013-03-25 02:35:59] VERBOSE[4744] chan_sip.c: --- (9 headers 0 lines) ---
[2013-03-25 02:36:09] VERBOSE[4854] app_dial.c: -- SIP/811-0000000b is making progress passing it to SIP/810-0000000a
[2013-03-25 02:36:09] VERBOSE[4854] chan_sip.c: Audio is at 11146
[2013-03-25 02:36:09] VERBOSE[4854] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2013-03-25 02:36:09] VERBOSE[4854] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2013-03-25 02:36:09] VERBOSE[4854] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2013-03-25 02:36:09] VERBOSE[4854] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2013-03-25 02:36:09] VERBOSE[4854] chan_sip.c:
<--- Transmitting (NAT) to MeraExtIP:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965277 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
<------------>
[2013-03-25 02:36:15] VERBOSE[4854] app_dial.c: -- SIP/811-0000000b answered SIP/810-0000000a
[2013-03-25 02:36:15] VERBOSE[4854] chan_sip.c: Audio is at 11146
[2013-03-25 02:36:15] VERBOSE[4854] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2013-03-25 02:36:15] VERBOSE[4854] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2013-03-25 02:36:15] VERBOSE[4854] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2013-03-25 02:36:15] VERBOSE[4854] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2013-03-25 02:36:15] VERBOSE[4854] chan_sip.c:
<--- Reliably Transmitting (NAT) to MeraExtIP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
<------------>
[2013-03-25 02:36:16] VERBOSE[4744] chan_sip.c: Retransmitting #1 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:17] VERBOSE[4744] chan_sip.c: Retransmitting #2 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:18] VERBOSE[4854] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/810-0000000a", "") in new stack
[2013-03-25 02:36:18] VERBOSE[4854] features.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/810-0000000a'
[2013-03-25 02:36:18] VERBOSE[4854] pbx.c: == Spawn extension (from-internal, 81189*********, 1) exited non-zero on 'SIP/810-0000000a'
[2013-03-25 02:36:18] VERBOSE[4854] chan_sip.c: Scheduling destruction of SIP dialog '40a42500ce7f4f10800000304884b7e0@mera' in 32000 ms (Method: INVITE)
[2013-03-25 02:36:19] VERBOSE[4744] chan_sip.c: Retransmitting #3 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:23] VERBOSE[4744] chan_sip.c: Retransmitting #4 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:27] VERBOSE[4744] chan_sip.c: Retransmitting #5 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:31] VERBOSE[4744] chan_sip.c: Retransmitting #6 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:35] VERBOSE[4744] chan_sip.c: Retransmitting #7 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:39] VERBOSE[4744] chan_sip.c: Retransmitting #8 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:43] VERBOSE[4744] chan_sip.c: Retransmitting #9 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:47] VERBOSE[4744] chan_sip.c: Retransmitting #10 (NAT) to MeraExtIP:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MeraExtIP:5060;branch=z9hG4bK-3aff2500ff7f4f10ff00003048ffffff;received=MeraExtIP;rport=5060
From: <sip:10007@MeraExtIP;user=phone>;tag=24ff2500ff7f4f10ff00003048ffffff
To: <sip:81189*********@AsteriskExtIP:5060;user=phone>;tag=as47c0779f
Call-ID: 40a42500ce7f4f10800000304884b7e0@mera
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:81189*********@AsteriskExtIP:5060>
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 1973965277 1973965278 IN IP4 AsteriskExtIP
s=Asterisk PBX 1.8.19.1
c=IN IP4 AsteriskExtIP
t=0 0
m=audio 11146 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
---
[2013-03-25 02:36:47] WARNING[4744] chan_sip.c: Retransmission timeout reached on transmission 40a42500ce7f4f10800000304884b7e0@mera for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[2013-03-25 02:36:47] VERBOSE[4744] chan_sip.c: Really destroying SIP dialog '40a42500ce7f4f10800000304884b7e0@mera' Method: INVITE
[2013-03-25 02:36:58] VERBOSE[4853] asterisk.c: -- Remote UNIX connection disconnected
===========rtp set debug on
[2013-03-25 02:21:32] VERBOSE[4743] asterisk.c: -- Remote UNIX connection disconnected
[2013-03-25 02:28:21] VERBOSE[4744] netsock2.c: == Using SIP RTP TOS bits 184
[2013-03-25 02:28:21] VERBOSE[4744] netsock2.c: == Using SIP RTP CoS mark 5
[2013-03-25 02:28:21] VERBOSE[4829] pbx.c: -- Executing [81189*********@from-internal:1] Dial("SIP/810-00000006", "SIP/811/*189*********") in new stack
[2013-03-25 02:28:21] VERBOSE[4829] netsock2.c: == Using SIP RTP TOS bits 184
[2013-03-25 02:28:21] VERBOSE[4829] netsock2.c: == Using SIP RTP CoS mark 5
[2013-03-25 02:28:21] VERBOSE[4829] app_dial.c: -- Called SIP/811/*189*********
[2013-03-25 02:28:21] VERBOSE[4829] app_dial.c: -- SIP/811-00000007 is ringing
[2013-03-25 02:28:30] DEBUG[4829] res_rtp_asterisk.c: RTP NAT: Got audio from other end. Now sending to address GWLOCALIP:16384
[2013-03-25 02:28:30] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024581, ts 262224, len 000160)
[2013-03-25 02:28:30] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024582, ts 262384, len 000160)
[2013-03-25 02:28:30] VERBOSE[4829] app_dial.c: -- SIP/811-00000007 is making progress passing it to SIP/810-00000006
[2013-03-25 02:28:30] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024583, ts 262544, len 000160)
[2013-03-25 02:28:30] VERBOSE[4829] res_rtp_asterisk.c: Sent RTP packet to MeraExtIP:28600 (type 08, seq 060865, ts 262544, len 000160)
[2013-03-25 02:28:30] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024584, ts 262704, len 000160)
[2013-03-25 02:28:30] VERBOSE[4829] res_rtp_asterisk.c: Sent RTP packet to MeraExtIP:28600 (type 08, seq 060866, ts 262704, len 000160)
[2013-03-25 02:28:30] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024585, ts 262864, len 000160)
......
[2013-03-25 02:28:32] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024651, ts 393456, len 000160)
[2013-03-25 02:28:32] VERBOSE[4829] res_rtp_asterisk.c: Sent RTP packet to MeraExtIP:28600 (type 08, seq 060933, ts 393456, len 000160)
[2013-03-25 02:28:32] VERBOSE[4829] app_dial.c: -- SIP/811-00000007 is making progress passing it to SIP/810-00000006
[2013-03-25 02:28:32] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024652, ts 393616, len 000160)
[2013-03-25 02:28:32] VERBOSE[4829] res_rtp_asterisk.c: Sent RTP packet to MeraExtIP:28600 (type 08, seq 060934, ts 393616, len 000160)
[2013-03-25 02:28:32] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024653, ts 393776, len 000160)
......
[2013-03-25 02:28:33] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024708, ts 402576, len 000160)
[2013-03-25 02:28:33] VERBOSE[4829] res_rtp_asterisk.c: Sent RTP packet to MeraExtIP:28600 (type 08, seq 060990, ts 402576, len 000160)
[2013-03-25 02:28:33] VERBOSE[4829] res_rtp_asterisk.c: Got RTP packet from GWLOCALIP:16384 (type 08, seq 024709, ts 402736, len 000160)
[2013-03-25 02:28:33] VERBOSE[4829] res_rtp_asterisk.c: Sent RTP packet to MeraExtIP:28600 (type 08, seq 060991, ts 402736, len 000160)
[2013-03-25 02:28:33] VERBOSE[4744] chan_sip.c: -- Got SIP response 486 "Busy Here" back from GWLOCALIP:5060
[2013-03-25 02:28:33] VERBOSE[4829] app_dial.c: -- SIP/811-00000007 is busy
[2013-03-25 02:28:33] VERBOSE[4829] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)
[2013-03-25 02:28:33] VERBOSE[4829] pbx.c: -- Executing [81189*********@from-internal:2] Hangup("SIP/810-00000006", "") in new stack
[2013-03-25 02:28:33] VERBOSE[4829] pbx.c: == Spawn extension (from-internal, 81189*********, 2) exited non-zero on 'SIP/810-00000006'
[2013-03-25 02:28:33] VERBOSE[4829] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/810-00000006", "") in new stack
[2013-03-25 02:28:33] VERBOSE[4829] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/810-00000006'
[2013-03-25 02:29:05] WARNING[4744] chan_sip.c: Retransmission timeout reached on transmission 4a178a00037e4f10800000304884b7e0@mera for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Retransmission timeout reached on transmission 5d8aaf0e12f15609521451180572ff15 for seqno 102 Такая проблема, какие варианты?
ну вы сами не видите по вашему логу что у вас transmit есть а receive(ответов) нету?
скорее всего фаервол, но точно сказать нельзя ибо вы ж редактировали логи.
Задан: 2013-03-25 02:48:43 +0400
Просмотрен: 13,742 раз
Обновлен: Mar 11 '16
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.
вынесите конфиги и логи в http://pastebin.com/, пожалуйста, невозжможно же читать - никто читать и не станет...
viktorkho ( 2013-03-25 10:08:44 +0400 )редактировать2 meral.
Нет, не вижу, где и как должны быть видны ответа receive? SIp или RTP?
sharkowolf ( 2013-03-25 13:48:24 +0400 )редактироватьчитайте goo.gl/THxFM. посмотрите тут этих трейсов НОРМАЛЬНЫХ вагоны. да должны быть в том же логе ответы.
meral ( 2013-03-25 13:52:51 +0400 )редактироватьhttp://asterisk-support.ru/question/38386/kak-poniat-chto-proiskhodit-na-asteriske/
meral ( 2013-03-25 13:53:40 +0400 )редактироватьСпасибо за ссылку, но проблема осталась. Какие еще будут рекомендации кроме как читать мануалы?
sharkowolf ( 2013-03-25 14:13:29 +0400 )редактироватьнанять специалиста который может настроить фаервол в линуксе и пробросить порт внутрь.
meral ( 2013-03-25 14:44:29 +0400 )редактироватьmeral, благодарю за очередной конкретный ответ. service iptables stop результата не дает. правила ниже также результата не дают:
iptables -t nat -A PREROUTING -i eth0 -p udp -m multiport --dport 5060,5061,10000:20000 -j DNAT --to-destination 172.0.1.30 iptables -A FORWARD -p udp -m multiport --dport 5060,5061,10000:20000 -i eth0 -j ACCEPT iptables -t nat -A PREROUTING -i eth0 -p udp -m multiport --dport 5060,5061,10000:20000 -j DNAT --to-destination 172.0.1.30 iptables -A FORWARD -o eth0 -m udp -p udp -s 192.168.1.3 -j ACCEPT iptables -A FORWARD -p udp -m multiport --dport 5060,5061,10000:20000 -d 172.0.1.30 -j ACCEPT
Какие еще варианты?
sharkowolf ( 2013-03-26 11:23:24 +0400 )редактироватьtcpdump udp port 5060. и смотрите приходят ли пакетики.
meral ( 2013-03-26 13:05:02 +0400 )редактироватьПакеты проходят. Если кому-то поможет: Обмен с mera * с чужим: Обмен mera с * моим: -> Invite -> Invite
sharkowolf ( 2013-03-26 13:57:31 +0400 )редактировать<- 100 <- 100 <- 183 <- 180 <- 200 -> Ack <- Bye решение - отвечать не 180 а сначала 183. В данный момент тестируем.
Проблема решена. В диал план пишем: в sip.conf
prematuremedia = no progressinband = never далее в диалплане в любом месте перед вызовом, хоть перед Dial:
[context] exten => _X.,1,Progress exten => _X.,n,*** как указано здесь: http://asterisk-support.ru/question/15697/posylka-sip-183-progress-bez-predshestvuiushchego/
sharkowolf ( 2013-03-27 01:00:54 +0400 )редактировать