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Asterisk 7911 Retransmission timeout

0

Добрый день!

Зашел в тупик с решением проблемы.

Есть Asterisk в локальной сети, NAT не используется.

К нему в подключены SPA303 и 7911G (SIP firmware 8-5-4S)

Если звоним с SPA 303 на 7911 - то проблем нет, все отрабатывает корректно.

А вот при звонке с 7911 на любые номер, будь то SPA или 7911 или выход по SIP в город, то звонки обрываются звонки через 32 сек. по таймауту. Проиходит это не регулярно, т.е. пару звонков на один и тот же эксеншен могут пройти нормально, а следующие 3 обрываться.

Asterisk SIP.CONF

[general]
language=ru
canreinvite=no
rtpholdtimeout=300
rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes
ignorereregexpire=yes
bindport=5060
bindaddr=0.0.0.0
sendrpid=yes

[5004]
callerid="ТЕСТ"
username=5004
secret=XXXXX
type=friend
host=dynamic
port=5060
context=internal
dtmfmode=rfc2833
nat=no
canreinvite=no
qualify=no

SIP DEBUG без ошибки:

root@ttn-sr-asx01:/etc/asterisk# asterisk -rvvvvvvv
Asterisk 11.13.1~dfsg-2~bpo70+1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.13.1~dfsg-2~bpo70+1 currently running on ttn-sr-asx01 (pid = 19685)
       > Saved useragent "Cisco/SPA303-7.6.1" for peer 5060
Sending to 10.10.20.56:5060 (no NAT)

<--- Transmitting (no NAT) to 10.10.20.56:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK674f1444;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 103 BYE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Sending to 10.10.20.56:5060 (no NAT)
Sending to 10.10.20.56:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 10.10.20.56:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK4487fe3c;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb0000584315006-4504e154
To: <sip:5001@10.10.20.1>;tag=as0318fcb3
Call-ID: c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56
CSeq: 101 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d7a8c46"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56' in 32000 ms (Method: INVITE)
Sending to 10.10.20.56:5060 (no NAT)
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24), peer - audio=(ulaw|alaw|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.20.56:28554
Looking for 5001 in internal (domain 10.10.20.1)
list_route: hop: <sip:5004@10.10.20.56:5060;transport=udp>

<--- Transmitting (no NAT) to 10.10.20.56:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK85475f3e;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb0000584315006-4504e154
To: <sip:5001@10.10.20.1>
Call-ID: c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Content-Length: 0


<------------>
    -- Executing [5001@internal:1] Verbose("SIP/5004-00000049", "1,Extension 5001") in new stack
 Extension 5001
    -- Executing [5001@internal:2] Set("SIP/5004-00000049", "CONNECTEDLINE(number,i)=5001") in new stack
    -- Executing [5001@internal:3] Set("SIP/5004-00000049", "RCIDNAME=Войтов Г.Н.") in new stack
    -- Executing [5001@internal:4] Set("SIP/5004-00000049", "CONNECTEDLINE(name)=Войтов Г.Н.") in new stack
    -- Executing [5001@internal:5] Set("SIP/5004-00000049", "CONNECTEDLINE(pres)=allowed") in new stack
    -- Executing [5001@internal:6] Answer("SIP/5004-00000049", "") in new stack
Audio is at 15882
Adding codec 100003 (ulaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100010 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.10.20.56:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK85475f3e;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb0000584315006-4504e154
To: <sip:5001@10.10.20.1>;tag=as1298d51e
Call-ID: c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 491548733 491548733 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 15882 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
       > 0x7fc004023db0 -- Probation passed - setting RTP source address to 10.10.20.56:28554
    -- Executing [5001@internal:7] Dial("SIP/5004-00000049", "SIP/5001,20,tT") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/5001
set_destination: Parsing <sip:5004@10.10.20.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.10.20.56:5060
Reliably Transmitting (no NAT) to 10.10.20.56:5060:
UPDATE sip:5004@10.10.20.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1:5060;branch=z9hG4bK56b77550
Max-Forwards: 70
From: <sip:5001@10.10.20.1>;tag=as1298d51e
To: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb0000584315006-4504e154
Contact: <sip:5001@10.10.20.1:5060>
Call-ID: c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56
CSeq: 102 UPDATE
User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
X-Asterisk-rpid-update: Yes
Content-Length: 0


---
    -- SIP/5001-0000004a is ringing
set_destination: Parsing <sip:5004@10.10.20.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.10.20.56:5060
Audio is at 15882
Adding codec 100003 (ulaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100010 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.20.56:5060:
INVITE sip:5004@10.10.20.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1:5060;branch=z9hG4bK7f35fa36
Max-Forwards: 70
From: <sip:5001@10.10.20.1>;tag=as1298d51e
To: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb0000584315006-4504e154
Contact: <sip:5001@10.10.20.1:5060>
Call-ID: c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 491548733 491548734 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 15882 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- SIP/5001-0000004a answered SIP/5004-00000049
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.20.56:28554
set_destination: Parsing <sip:5004@10.10.20.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.10.20.56:5060
Transmitting (no NAT) to 10.10.20.56:5060:
ACK sip:5004@10.10.20.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1:5060;branch=z9hG4bK7a18d622
Max-Forwards: 70
From: <sip:5001@10.10.20.1>;tag=as1298d51e
To: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb0000584315006-4504e154
Contact: <sip:5001@10.10.20.1:5060>
Call-ID: c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Content-Length: 0


---
       > 0x7fc004023db0 -- Probation passed - setting RTP source address to 10.10.20.56:28554
       > 0x7fc00406ef50 -- Probation passed - setting RTP source address to 10.10.20.20:16502
[Oct 23 16:19:13] WARNING[19702]: chan_sip.c:4028 retrans_pkt: Retransmission timeout reached on transmission 108ccfe0-108d000e-dc4bd9a3-7402d675@10.10.20.29 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
       > Saved useragent "Cisco/SPA303-7.6.1" for peer 5041
[Oct 23 16:19:28] NOTICE[19702]: chan_sip.c:14995 sip_reregister:    -- Re-registration for  356931@195.239.164.48
[Oct 23 16:19:28] NOTICE[19702]: chan_sip.c:23511 handle_response_register: Outbound Registration: Expiry for 195.239.164.48 is 120 sec (Scheduling reregistration in 105 s)
[Oct 23 16:19:28] NOTICE[19702]: chan_sip.c:14995 sip_reregister:    -- Re-registration for  356932@195.239.164.48
[Oct 23 16:19:28] NOTICE[19702]: chan_sip.c:23511 handle_response_register: Outbound Registration: Expiry for 195.239.164.48 is 120 sec (Scheduling reregistration in 105 s)
  == Spawn extension (internal, 5001, 7) exited non-zero on 'SIP/5004-00000049'
Scheduling destruction of SIP dialog 'c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:5004@10.10.20.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.10.20.56:5060
Reliably Transmitting (no NAT) to 10.10.20.56:5060:
BYE sip:5004@10.10.20.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1:5060;branch=z9hG4bK3d90fd52
Max-Forwards: 70
From: <sip:5001@10.10.20.1>;tag=as1298d51e
To: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb0000584315006-4504e154
Call-ID: c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56
CSeq: 104 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Proxy-Authorization: Digest username="5004", realm="asterisk", algorithm=MD5, uri="sip:10.10.20.1", nonce="5d7a8c46", response="62c7cebe44fa09c0c0ddaaea7daba48f"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Really destroying SIP dialog 'c0626b62-bcb00005-b4b41d8e-df05e4ac@10.10.20.56' Method: ACK
ttn-sr-asx01*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

SIP DEBUG с ошибкой:

Asterisk 11.13.1~dfsg-2~bpo70+1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.13.1~dfsg-2~bpo70+1 currently running on ttn-sr-asx01 (pid = 19685)
ttn-sr-asx01*CLI> sip set debug peer 5004
SIP Debugging Enabled for IP: 10.10.20.56
Sending to 10.10.20.56:5060 (no NAT)
Sending to 10.10.20.56:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 10.10.20.56:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK08e3a172;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as659bec4b
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 101 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f28dfd3"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56' in 32000 ms (Method: INVITE)
Sending to 10.10.20.56:5060 (no NAT)
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24), peer - audio=(ulaw|alaw|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.20.56:25534
Looking for 5001 in internal (domain 10.10.20.1)
list_route: hop: <sip:5004@10.10.20.56:5060;transport=udp>

<--- Transmitting (no NAT) to 10.10.20.56:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Content-Length: 0


<------------>
    -- Executing [5001@internal:1] Verbose("SIP/5004-00000043", "1,Extension 5001") in new stack
 Extension 5001
    -- Executing [5001@internal:2] Set("SIP/5004-00000043", "CONNECTEDLINE(number,i)=5001") in new stack
    -- Executing [5001@internal:3] Set("SIP/5004-00000043", "RCIDNAME=Войтов Г.Н.") in new stack
    -- Executing [5001@internal:4] Set("SIP/5004-00000043", "CONNECTEDLINE(name)=Войтов Г.Н.") in new stack
    -- Executing [5001@internal:5] Set("SIP/5004-00000043", "CONNECTEDLINE(pres)=allowed") in new stack
    -- Executing [5001@internal:6] Answer("SIP/5004-00000043", "") in new stack
Audio is at 14766
Adding codec 100003 (ulaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100010 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.10.20.56:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
       > 0x7fc004049b70 -- Probation passed - setting RTP source address to 10.10.20.56:25534
    -- Executing [5001@internal:7] Dial("SIP/5004-00000043", "SIP/5001,20,tT") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/5001
set_destination: Parsing <sip:5004@10.10.20.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.10.20.56:5060
Reliably Transmitting (no NAT) to 10.10.20.56:5060:
UPDATE sip:5004@10.10.20.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1:5060;branch=z9hG4bK124ec497
Max-Forwards: 70
From: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
To: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
Contact: <sip:5001@10.10.20.1:5060>
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 UPDATE
User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
X-Asterisk-rpid-update: Yes
Content-Length: 0


---
    -- SIP/5001-00000044 is ringing
Retransmitting #1 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
set_destination: Parsing <sip:5004@10.10.20.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.10.20.56:5060
Reliably Transmitting (no NAT) to 10.10.20.56:5060:
UPDATE sip:5004@10.10.20.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.20.1:5060;branch=z9hG4bK582921ba
Max-Forwards: 70
From: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
To: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
Contact: <sip:5001@10.10.20.1:5060>
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 103 UPDATE
User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
X-Asterisk-rpid-update: Yes
Content-Length: 0


---
    -- SIP/5001-00000044 answered SIP/5004-00000043
       > 0x2b158c0 -- Probation passed - setting RTP source address to 10.10.20.20:16500
Retransmitting #2 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #3 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #4 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
       > Saved useragent "Cisco/SPA303-7.6.1" for peer 5061
Retransmitting #5 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #6 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #7 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #8 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #9 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #10 (no NAT) to 10.10.20.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.20.56:5060;branch=z9hG4bK66ac9dc2;received=10.10.20.56
From: "5004" <sip:5004@10.10.20.1>;tag=c0626b62bcb00004a1c12de2-f7e93e12
To: <sip:5001@10.10.20.1>;tag=as0e3bb7c6
Call-ID: c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@10.10.20.1:5060>
Remote-Party-ID: "Войтов Г.Н." <sip:5001@10.10.20.1>;party=called;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 729299618 729299618 IN IP4 10.10.20.1
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 10.10.20.1
t=0 0
m=audio 14766 RTP/AVP 0 0 8 8 18 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Oct 23 16:17:29] WARNING[19702]: chan_sip.c:4028 retrans_pkt: Retransmission timeout reached on transmission c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Oct 23 16:17:29] WARNING[19702]: chan_sip.c:4057 retrans_pkt: Hanging up call c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (internal, 5001, 7) exited non-zero on 'SIP/5004-00000043'
Really destroying SIP dialog 'c0626b62-bcb00004-c4971102-540a01b2@10.10.20.56' Method: INVITE
ttn-sr-asx01*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).

Не нашел как комментировать Ваш ответ, так что пишу здесь Обновил на телефонах прошивку до последней на сайте Cisco SIP11.9-4-2SR1-1S

Пришлось подкорректировать конфиги телефонов Проблем осталась

.

.

.

. Вопрос вроде решен. Выставил следующие параметры:

sendrpid=no
rpid_update=no

Перестал приходить UPDATE и перестали валиться сессии.

Как будет свободное время, сделаю, что бы параметр Remote-Party-ID устанавливался через оператор.

Спасибо товарищу meral, направил в нужную сторону

Ссылка на топ, который помог:

http://forums.digium.com/viewtopic.php?p=164742

удалить закрыть спам изменить тег редактировать

спросил 2015-10-27 14:39:56 +0400

scip Gravatar scip
11 1 3

обновил 2015-10-30 12:34:41 +0400

Comments

теория - одна из сторон (обычно asterisk завершает вызов) - а почему ? а потому что не видит RTP трафика и считает что канал сдох - причин много и одна из них включен VAD на телефоне или трафик идет на прямую в обход астера - на астере = directmedia=no

awsswa ( 2015-10-27 15:49:12 +0400 )редактировать

Поставил глобально directmedia=no, проблема осталась

А что касается VAD, полагаю обрывы должны происходить в момент затишья. Но обрывает в момент разговора. И tcpdump показывает, что RTP не прекращается.

scip ( 2015-10-28 09:17:49 +0400 )редактировать

снимайте дамп и файл pcap бросайте мне = снимайте !!!

awsswa ( 2015-10-28 14:23:30 +0400 )редактировать

причем тут rtp трафик? вызов завершается ибо нет ACK. есть ли при этом rtp - неважно,

meral ( 2015-10-30 06:16:07 +0400 )редактировать

1 Ответ

1

у вас во втором случае в дебаге UPDATE, на который астериск в соответсвии с RFC шлет OK, ну и на OK нет ACK, соответсвенно делается ретрансмиты а когда все ретрансмиты неудачны астериск завершает звонок.

почему так происходит? неизвестно. возможно глюк прошивки того из девайсов с которым это происходит.

как минимум цисковская прошивка не является последней, перепрошейте.

ссылка удалить спам редактировать

ответил 2015-10-28 14:52:57 +0400

meral Gravatar meral flag of Ukraine
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опа - невнимательно смотрел оказывается - может вообще выключить ? sendrpid=no

awsswa ( 2015-10-28 15:05:31 +0400 )редактировать

вам решать. я бы сначала проапдейтился.

meral ( 2015-10-29 06:03:13 +0400 )редактировать

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Задан: 2015-10-27 14:39:56 +0400

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