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Retransmission timeout reached on transmission

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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
удалить закрыть спам изменить тег редактировать

спросил 2013-03-25 02:48:43 +0400

sharkowolf Gravatar sharkowolf
1 2 1 2

обновил 2013-03-25 13:26:16 +0400

meral Gravatar meral flag of Ukraine
21228 23 18 169
http://pro-sip.net/

Comments

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 )редактировать

2 meral.

Нет, не вижу, где и как должны быть видны ответа receive? SIp или RTP?

sharkowolf ( 2013-03-25 13:48:24 +0400 )редактировать

читайте 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 )редактировать

Спасибо за ссылку, но проблема осталась. Какие еще будут рекомендации кроме как читать мануалы?

sharkowolf ( 2013-03-25 14:13:29 +0400 )редактировать

нанять специалиста который может настроить фаервол в линуксе и пробросить порт внутрь.

meral ( 2013-03-25 14:44:29 +0400 )редактировать

meral, 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 )редактировать

tcpdump udp port 5060. и смотрите приходят ли пакетики.

meral ( 2013-03-26 13:05:02 +0400 )редактировать

Пакеты проходят. Если кому-то поможет: Обмен с mera * с чужим: Обмен mera с * моим: -> Invite -> Invite
<- 100 <- 100 <- 183 <- 180 <- 200 -> Ack <- Bye решение - отвечать не 180 а сначала 183. В данный момент тестируем.

sharkowolf ( 2013-03-26 13:57:31 +0400 )редактировать

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

2 Ответа

0

Retransmission timeout reached on transmission 5d8aaf0e12f15609521451180572ff15 for seqno 102 Такая проблема, какие варианты?

ссылка удалить спам редактировать

ответил 2016-03-11 05:35:09 +0400

hrapp Gravatar hrapp
1 1 1
0

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.

ссылка удалить спам редактировать

ответил 2013-03-25 13:28:06 +0400

meral Gravatar meral flag of Ukraine
21228 23 18 169
http://pro-sip.net/

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Задан: 2013-03-25 02:48:43 +0400

Просмотрен: 6,028 раз

Обновлен: Mar 11

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.