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спросил 2013-04-25 11:08:36 +0400

AlexiusFlavius Gravatar AlexiusFlavius

Проблема с музыкой и односторонний звук после удержания звонка

Всем день добрый,

Перво на перво я использую астериск 1.8.21 на CentOS 6.4, сборка PIAF Есть две проблемы следующего характера: К серверу подключены 3 телефона: snom 300, snom 320 и grandstream 2010 Между ними разговоры идут используя SRTP и TLS, все нормально дозваниваются и говорят Теперь если на сноме использовать кнопку ХОЛД то должна быть музыка на удержании, но вместо нее тишина, В CLI я вижу следующее:

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c:

<--- SIP read from TLS:172.17.0.186:2059 --->

INVITE sip:999997@172.17.3.241:5061;transport=TLS SIP/2.0

Via: SIP/2.0/TLS 172.17.0.186:2059;branch=z9hG4bK-7nze9as0gj61;rport

From: "999999" <sip:999999@172.17.3.241>;tag=yk1iamm3ef

To: <sip:999997@172.17.3.241;user=phone>;tag=as1ff254e3

Call-ID: 3c2a472f4b22-ixhjamkj85m0

CSeq: 5 INVITE

Max-Forwards: 70

Contact: <sip:999999@172.17.0.186:2059;transport=tls;line=1n0coz8i>;reg-id=1

P-Key-Flags: keys="3"

User-Agent: snom320/7.3.30

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Content-Type: application/sdp

Content-Length: 367

v=0

o=root 10291801 10291804 IN IP4 172.17.0.186

s=call

c=IN IP4 172.17.0.186

t=0 0

m=audio 65418 RTP/SAVP 0 8 9 99 3 18 4 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:99 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendonly

<------------->

[2013-04-25 02:20:49] VERBOSE[2155]

chan_sip.c: --- (18 headers 17 lines)

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Sending to 172.17.0.186:2059 (no NAT)

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 0

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 8

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 9

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 0

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 8

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 9

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 99

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 3

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 18

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 4

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 101

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found RTP audio format 101

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format pcmu for ID 0

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format pcma for ID 8

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format pcma for ID 8

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format g722 for ID 9

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format g726-32 for ID 99

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format gsm for ID 3

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format gsm for ID 3

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format g729 for ID 18

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format g723 for ID 4

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format telephone-event for ID 101

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c: Found audio description format telephone-event for ID 101

[2013-04-25 02:20:49] WARNING[2155] chan_sip.c: Rejecting secure audio stream without encryption details: audio 65418 RTP/SAVP 0 8 9 99 3 18 4 101

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c:

<--- Reliably Transmitting (no NAT) to 172.17.0.186:2059 --->

SIP/2.0 488 Not acceptable here

Via: SIP/2.0/TLS 172.17.0.186:2059;branch=z9hG4bK-7nze9as0gj61;received=172.17.0.186; rport=2059

From: "999999" <sip:999999@172.17.3.241>;tag=yk1iamm3ef

To: <sip:999997@172.17.3.241;user=phone>;tag=as1ff254e3

Call-ID: 3c2a472f4b22-ixhjamkj85m0

CSeq: 5 INVITE

Server: FPBX-2.9.0(1.8.21.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

Content-Length: 0

Тишина из за реджекта SRTP, но я не могу понять как это исправить, может кто сталкивался, Кстати если HOLD запустить на гранстриме то все работает и на другом конце идет музыка. Тут есть еще одна проблема, если разговор длится более 8 минут, то если запустить HOLD, а потом отменить, то слышен либо шум, вместо разговора, либо тишина и разговор приходиться прерывать, в CLI видно следующее:

[2013-04-25 02:35:45] VERBOSE[1721] chan_sip.c:

<--- SIP read from UDP:172.17.3.143:5061 --->

ACK sip:999999@172.17.3.241:5060 SIP/2.0

Via: SIP/2.0/UDP 172.17.3.143:5061;branch=z9hG4bKa3c44852dabd7bad

From: <sip:999997@172.17.3.143:5061;transport=udp>;tag=66a86b21e19fda81

To: "test-exten" <sip:999999@172.17.3.241>;tag=as43749bcc

Contact: <sip:999997@172.17.3.143:5061;transport=udp>

Supported: path

X-Grandstream-PBX: true

Call-ID: 3816661d337f21613f1c822625df9e9c@172.17.3.241:5060

CSeq: 32723 ACK

User-Agent: Grandstream GXP2010 1.2.5.3

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE

Content-Length: 0

<------------->

[2013-04-25 02:35:45] VERBOSE[1721]

chan_sip.c: --- (13 headers 0 lines)

[2013-04-25 02:35:45] WARNING[7912] res_srtp.c: SRTP unprotect failed with: authentication failure 10

[2013-04-25 02:35:45] VERBOSE[1721] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.3.184:2054:

OPTIONS sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb SIP/2.0

Via: SIP/2.0/TLS 172.17.3.241:5061;branch=z9hG4bK01ca8748

Max-Forwards: 70

From: "Unknown" <sip:unknown@172.17.3.241>;tag=as2e3228c2

To: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb>

Contact: <sip:unknown@172.17.3.241:5061;transport=tls>

Call-ID: 438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061

CSeq: 102 OPTIONS

User-Agent: FPBX-2.9.0(1.8.21.0)

Date: Thu, 25 Apr 2013 06:35:45 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

[2013-04-25 02:35:45] VERBOSE[3272] chan_sip.c:

<--- SIP read from TLS:172.17.3.184:2054 --->

SIP/2.0 200 OK

Via: SIP/2.0/TLS 172.17.3.241:5061;branch=z9hG4bK01ca8748

From: "Unknown" <sip:unknown@172.17.3.241>;tag=as2e3228c2

To: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb>

Call-ID: 438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061

CSeq: 102 OPTIONS

Contact: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb>;reg-id=1

User-Agent: snom300/7.3.30

Accept-Language: en

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Content-Length: 0

<------------->

[2013-04-25 02:35:45] VERBOSE[3272]

chan_sip.c: --- (14 headers 0 lines)

[2013-04-25 02:35:46] VERBOSE[1721] chan_sip.c: Really destroying SIP dialog '438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061' Method: OPTIONS

[2013-04-25 02:35:47] WARNING[7912] res_srtp.c: SRTP unprotect failed with: authentication failure 110

[2013-04-25 02:35:47] VERBOSE[1721] chan_sip.c:

<--- SIP read from UDP:172.17.3.250:5060 --->

OPTIONS sip:172.17.3.241 SIP/2.0

Via: SIP/2.0/UDP 172.17.3.250:5060;branch=z9hG4bK1555d74f;rport

Max-Forwards: 70

From: "Unknown" <sip:unknown@172.17.3.250>;tag=as1bb1be7a

To: <sip:172.17.3.241>

Contact: <sip:unknown@172.17.3.250:5060>

Call-ID: 3bc7d30056bbf39c7550c13b1d6f9b97@172.17.3.250:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-2.9.0(1.8.15.1)

Date: Thu, 25 Apr 2013 06:58:01 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

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No.2 Revision
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обновил 2013-04-25 12:22:20 +0400

zzuz Gravatar zzuz flag of Russian Federation

http://line24.ru/

Проблема с музыкой и односторонний звук после удержания звонка

Всем день добрый,

Перво на перво я использую астериск 1.8.21 на CentOS 6.4, сборка PIAF Есть две проблемы следующего характера: К серверу подключены 3 телефона: snom 300, snom 320 и grandstream 2010 Между ними разговоры идут используя SRTP и TLS, все нормально дозваниваются и говорят Теперь если на сноме использовать кнопку ХОЛД то должна быть музыка на удержании, но вместо нее тишина, В CLI я вижу следующее:

[2013-04-25 02:20:49] VERBOSE[2155] chan_sip.c:

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c:
<--- SIP read from
 TLS:172.17.0.186:2059 --->
 
--->
INVITE
 sip:999997@172.17.3.241:5061;transport=TLS
  SIP/2.0
 
SIP/2.0
Via: SIP/2.0/TLS
  172.17.0.186:2059;branch=z9hG4bK-7nze9as0gj61;rport
 
172.17.0.186:2059;branch=z9hG4bK-7nze9as0gj61;rport
From: "999999"
  <sip:999999@172.17.3.241>;tag=yk1iamm3ef
 
<sip:999999@172.17.3.241>;tag=yk1iamm3ef
To:
  <sip:999997@172.17.3.241;user=phone>;tag=as1ff254e3
 
<sip:999997@172.17.3.241;user=phone>;tag=as1ff254e3
Call-ID: 3c2a472f4b22-ixhjamkj85m0
 
3c2a472f4b22-ixhjamkj85m0
CSeq: 5 INVITE
 
INVITE
Max-Forwards: 70
 
70
Contact:
  <sip:999999@172.17.0.186:2059;transport=tls;line=1n0coz8i>;reg-id=1
 
<sip:999999@172.17.0.186:2059;transport=tls;line=1n0coz8i>;reg-id=1
P-Key-Flags: keys="3"
 
keys="3"
User-Agent: snom320/7.3.30
 
snom320/7.3.30
Accept: application/sdp
 
application/sdp
Allow: INVITE, ACK, CANCEL, BYE,
 REFER, OPTIONS, NOTIFY, SUBSCRIBE,
 PRACK, MESSAGE,  INFO
 
INFO
Allow-Events: talk, hold, refer,
  call-info
 
call-info
Supported: timer, 100rel, replaces,
  from-change
 
from-change
Session-Expires: 3600;refresher=uas
 
3600;refresher=uas
Min-SE: 90
 
90
Content-Type: application/sdp
 
application/sdp
Content-Length: 367
 
v=0
 
367
v=0
o=root 10291801 10291804 IN IP4
  172.17.0.186
 
s=call
 
172.17.0.186
s=call
c=IN IP4 172.17.0.186
 
172.17.0.186
t=0 0
 
0
m=audio 65418 RTP/SAVP 0 8 9 99 3 18 4
  101
 
101
a=rtpmap:0 pcmu/8000
 
pcmu/8000
a=rtpmap:8 pcma/8000
 
pcma/8000
a=rtpmap:9 g722/8000
 
g722/8000
a=rtpmap:99 g726-32/8000
 
g726-32/8000
a=rtpmap:3 gsm/8000
 
gsm/8000
a=rtpmap:18 g729/8000
 
g729/8000
a=rtpmap:4 g723/8000
 
g723/8000
a=rtpmap:101 telephone-event/8000
 
telephone-event/8000
a=fmtp:101 0-16
 
a=ptime:20
 
a=sendonly
 
<------------->
 
[2013-04-25 02:20:49] VERBOSE[2155]
 
0-16
a=ptime:20
a=sendonly

<------------->

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: --- (18 headers 17 lines)
 
[2013-04-25 02:20:49] VERBOSE[2155]
  lines)
---

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Sending to
 172.17.0.186:2059 (no NAT)
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 0
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 8
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 9
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 0
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 8
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 9
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 99
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 3
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 18
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 4
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 101
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found RTP audio format 101
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  NAT)

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 0

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 8

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 9

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 0

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 8

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 9

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 99

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 3

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 18

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 4

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 101

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found RTP audio format 101

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format pcmu  for ID 0
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  0

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format pcma  for ID 8
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  8

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format pcma  for ID 8
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  8

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format g722  for ID 9
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  9

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format g726-32  for ID 99
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  99

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format gsm for  ID 3
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  3

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format gsm for  ID 3
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  3

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format g729  for ID 18
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  18

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format g723  for ID 4
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  4

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format  telephone-event for ID 101
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c: Found audio description
  101

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c: Found audio description
format  telephone-event for ID 101
 
101

[2013-04-25 02:20:49]
 WARNING[2155] **WARNING[2155]** chan_sip.c: Rejecting secure audio stream  without
 encryption details: audio 65418
 RTP/SAVP 0 8 9 99 3 18 4 101
 
[2013-04-25 02:20:49] VERBOSE[2155]
  chan_sip.c:
 
101

[2013-04-25 02:20:49] VERBOSE[2155]
chan_sip.c:

<--- Reliably Transmitting (no NAT) to
 172.17.0.186:2059 --->
 
--->
SIP/2.0 488 Not acceptable here
 
here
Via: SIP/2.0/TLS
 172.17.0.186:2059;branch=z9hG4bK-7nze9as0gj61;received=172.17.0.186;
  rport=2059
 
rport=2059
From: "999999"
  <sip:999999@172.17.3.241>;tag=yk1iamm3ef
 
<sip:999999@172.17.3.241>;tag=yk1iamm3ef
To:
  <sip:999997@172.17.3.241;user=phone>;tag=as1ff254e3
 
<sip:999997@172.17.3.241;user=phone>;tag=as1ff254e3
Call-ID: 3c2a472f4b22-ixhjamkj85m0
 
3c2a472f4b22-ixhjamkj85m0
CSeq: 5 INVITE
 
INVITE
Server: FPBX-2.9.0(1.8.21.0)
 
FPBX-2.9.0(1.8.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS,
 BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
  PUBLISH
 
PUBLISH
Supported: replaces, timer
 
timer
Session-Expires: 1800;refresher=uas
 
1800;refresher=uas
X-Asterisk-HangupCause: Normal
  Clearing
 
Clearing

X-Asterisk-HangupCauseCode: 16
 
16

Content-Length: 0

0 Тишина из за реджекта SRTP, но я не могу понять как это исправить, может кто сталкивался, Кстати если HOLD запустить на гранстриме то все работает и на другом конце идет музыка. Тут есть еще одна проблема, если разговор длится более 8 минут, то если запустить HOLD, а потом отменить, то слышен либо шум, вместо разговора, либо тишина и разговор приходиться прерывать, в CLI видно следующее:

следующее: [2013-04-25 02:35:45] VERBOSE[1721] chan_sip.c:

chan_sip.c: <--- SIP read from UDP:172.17.3.143:5061 --->

---> ACK sip:999999@172.17.3.241:5060 SIP/2.0

SIP/2.0 Via: SIP/2.0/UDP 172.17.3.143:5061;branch=z9hG4bKa3c44852dabd7bad

172.17.3.143:5061;branch=z9hG4bKa3c44852dabd7bad From: <sip:999997@172.17.3.143:5061;transport=udp>;tag=66a86b21e19fda81

<sip:999997@172.17.3.143:5061;transport=udp>;tag=66a86b21e19fda81 To: "test-exten" <sip:999999@172.17.3.241>;tag=as43749bcc

<sip:999999@172.17.3.241>;tag=as43749bcc Contact: <sip:999997@172.17.3.143:5061;transport=udp>

<sip:999997@172.17.3.143:5061;transport=udp> Supported: path

path X-Grandstream-PBX: true

true Call-ID: 3816661d337f21613f1c822625df9e9c@172.17.3.241:5060

3816661d337f21613f1c822625df9e9c@172.17.3.241:5060 CSeq: 32723 ACK

ACK User-Agent: Grandstream GXP2010 1.2.5.3

1.2.5.3 Max-Forwards: 70

70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE

INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0

<------------->

0 <-------------> [2013-04-25 02:35:45] VERBOSE[1721]

VERBOSE[1721] chan_sip.c: --- (13 headers 0 lines)

lines) --- [2013-04-25 02:35:45] WARNING[7912] res_srtp.c: SRTP unprotect failed with: authentication failure 10

10 [2013-04-25 02:35:45] VERBOSE[1721] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.3.184:2054:

172.17.3.184:2054: OPTIONS sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb SIP/2.0

SIP/2.0 Via: SIP/2.0/TLS 172.17.3.241:5061;branch=z9hG4bK01ca8748

172.17.3.241:5061;branch=z9hG4bK01ca8748 Max-Forwards: 70

70 From: "Unknown" <sip:unknown@172.17.3.241>;tag=as2e3228c2

<sip:Unknown@172.17.3.241>;tag=as2e3228c2 To: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb>

<sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb> Contact: <sip:unknown@172.17.3.241:5061;transport=tls>

<sip:Unknown@172.17.3.241:5061;transport=TLS> Call-ID: 438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061

438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061 CSeq: 102 OPTIONS

OPTIONS User-Agent: FPBX-2.9.0(1.8.21.0)

FPBX-2.9.0(1.8.21.0) Date: Thu, 25 Apr 2013 06:35:45 GMT

GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

PUBLISH Supported: replaces, timer

timer Content-Length: 0

0 --- [2013-04-25 02:35:45] VERBOSE[3272] chan_sip.c:

chan_sip.c: <--- SIP read from TLS:172.17.3.184:2054 --->

---> SIP/2.0 200 OK

OK Via: SIP/2.0/TLS 172.17.3.241:5061;branch=z9hG4bK01ca8748

172.17.3.241:5061;branch=z9hG4bK01ca8748 From: "Unknown" <sip:unknown@172.17.3.241>;tag=as2e3228c2

<sip:Unknown@172.17.3.241>;tag=as2e3228c2 To: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb>

<sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb> Call-ID: 438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061

438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061 CSeq: 102 OPTIONS

OPTIONS Contact: <sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb>;reg-id=1

<sip:999998@172.17.3.184:2054;transport=tls;line=oi2k6svb>;reg-id=1 User-Agent: snom300/7.3.30

snom300/7.3.30 Accept-Language: en

en Accept: application/sdp

application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

INFO Allow-Events: talk, hold, refer, call-info

call-info Supported: timer, 100rel, replaces, from-change

from-change Content-Length: 0

<------------->

0 <-------------> [2013-04-25 02:35:45] VERBOSE[3272]

VERBOSE[3272] chan_sip.c: --- (14 headers 0 lines)

lines) --- [2013-04-25 02:35:46] VERBOSE[1721] chan_sip.c: Really destroying SIP dialog '438959a738d6c7aa349174e94bc5ee14@172.17.3.241:5061' Method: OPTIONS

OPTIONS [2013-04-25 02:35:47] WARNING[7912] res_srtp.c: SRTP unprotect failed with: authentication failure 110

110 [2013-04-25 02:35:47] VERBOSE[1721] chan_sip.c:

chan_sip.c: <--- SIP read from UDP:172.17.3.250:5060 --->

---> OPTIONS sip:172.17.3.241 SIP/2.0

SIP/2.0 Via: SIP/2.0/UDP 172.17.3.250:5060;branch=z9hG4bK1555d74f;rport

172.17.3.250:5060;branch=z9hG4bK1555d74f;rport Max-Forwards: 70

70 From: "Unknown" <sip:unknown@172.17.3.250>;tag=as1bb1be7a

<sip:Unknown@172.17.3.250>;tag=as1bb1be7a To: <sip:172.17.3.241>

<sip:172.17.3.241> Contact: <sip:unknown@172.17.3.250:5060>

<sip:Unknown@172.17.3.250:5060> Call-ID: 3bc7d30056bbf39c7550c13b1d6f9b97@172.17.3.250:5060

3bc7d30056bbf39c7550c13b1d6f9b97@172.17.3.250:5060 CSeq: 102 OPTIONS

OPTIONS User-Agent: FPBX-2.9.0(1.8.15.1)

FPBX-2.9.0(1.8.15.1) Date: Thu, 25 Apr 2013 06:58:01 GMT

GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

PUBLISH Supported: replaces, timer

timer Content-Length: 0

0
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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.