Новые ответы

Нужна помощь в настройке Cisco 1760 c 2FXO,

lockdoc, 27-Май-15, 22:56 [смотреть все]

Имеется cisco 1760 (64Мб) с платой 2FXO и PVDM-256K-4 (прошивка c1700-ipvoice-mz.123-23.bin).
Подробно:
------
Cisco Internetwork Operating System Software
IOS (tm) C1700 Software (C1700-IPVOICE-M), Version 12.3(23), RELEASE SOFTWARE (fc5)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2007 by cisco Systems, Inc.
Compiled Tue 24-Jul-07 14:47 by stshen
Image text-base: 0x8000816C, data-base: 0x81455D6C
ROM: System Bootstrap, Version 12.2(4r)XL, RELEASE SOFTWARE (fc1)
System returned to ROM by reload at 12:05:11 Moscow Wed May 27 2015
System restarted at 12:07:15 Moscow Wed May 27 2015
System image file is "flash:c1700-ipvoice-mz.123-23.bin"
cisco 1760 (MPC860P) processor (revision 0x200) with 59225K/6311K bytes of memory.
Processor board ID FOC071218L4 (1906716500), with hardware revision BB67
MPC860P processor: part number 5, mask 2
Bridging software.
X.25 software, Version 3.0.0.
1 Ethernet/IEEE 802.3 interface(s)
1 FastEthernet/IEEE 802.3 interface(s)
2 Voice FXO interface(s)
32K bytes of non-volatile configuration memory.
32768K bytes of processor board System flash (Read/Write)
-------
Необходимо подключить сей аппарат к Asterisk-у и добавить две телефонные линии в план телефонии с возможностью звонков в город (FXO) и прием звонков из города на ip-телефоны.
Долго бился, но так и не заставил cisco принимать входящие из города. Звонки в город идут нормально, а вот входящие из города на FXO сразу идет занято. По всей видимости cisco не может набрать voip номер (передать его asterisk-у).
Даже если отключить "connection plar 551" то cisco при звонке из города дает в телефон тон, набираю любой номер и таже занято...
Скорее всего где то я допустил ошибку?
Имеется следующий конфиг:
--------
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname cisco-2fxo
!
boot-start-marker
boot system flash:c1700-ipvoice-mz.123-23.bin
boot system flash:c1700-sv3y-mz.123-26.bin
boot-end-marker
!
enable secret 5 .....................
!
clock timezone Moscow 3
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 1
!
no aaa new-model
ip subnet-zero
ip cef
!
voice service pots
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711alaw
codec preference 3 g711ulaw
!
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
interface Ethernet0/0
no ip address
shutdown
half-duplex
!
interface FastEthernet0/0
ip address dhcp hostname cisco-2fxo
speed auto
!
ip classless
no ip http server
!
voice-port 1/0
supervisory disconnect dualtone mid-call
cptone RU
connection plar 551
impedance complex1
description tel 53-09-31
station-id name 53-09-31
caller-id enable
!
voice-port 1/1
supervisory disconnect dualtone mid-call
cptone RU
connection plar 552
impedance complex1
description tel 53-93-00
station-id name 53-93-00
!
dial-peer voice 1 pots
description tel 53-09-31
destination-pattern 703......
port 1/0
forward-digits 6
!
dial-peer voice 2 pots
description tel 53-93-00
destination-pattern 704......
port 1/1
forward-digits 6
!
dial-peer voice 10 voip
description all
destination-pattern 55[12]
voice-class codec 2
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
no vad
!
sip-ua
sip-server ipv4:192.168.51.13
!
!
line con 0
line aux 0
line vty 0 4
password .......
login
!
ntp clock-period 17208540
ntp server 195.2.64.6
ntp server 192.168.51.1
end
--------
Астериск имеет следующий конфиг
-----
[703]
type=friend
host=192.168.51.14
qualify=yes
qualifyfreq=10
disallow=all
allow=ulaw
allow=alaw
context=main
nat=no
insecury=invite
[704]
type=peer
host=192.168.51.14
qualify=yes
qualifyfreq=10
disallow=all
allow=ulaw
allow=alaw
context=main
nat=no
insecury=invite
---------
Звоню в город по следующему экстеншену
---------
exten => _XXXXXX,1,Dial(SIP/192.168.51.14/703${EXTEN},60)
exten => _XXXXXX,n,HangUp()
---------

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO,

ShyLion, 06:40 , 28-Май-15 (1)

debug ccsip messages
хотя не уверен что в 12.3 эта команда есть...

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO,

lockdoc, 08:14 , 28-Май-15 (2)

> debug ccsip messages
> хотя не уверен что в 12.3 эта команда есть...
Да такая команда есть, вот ответ - лог очень большой (выбрал лог только при: ждал 3 сек., потом звонок, потом еще 3 секунды).
--------------
cisco-2fxo#no debug ccsip messages
SIP Call messages tracing is disabled
cisco-2fxo#debug ccsip messages
SIP Call messages tracing is enabled
cisco-2fxo#
May 28 05:10:42.818: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK3119996f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as4617a5bd
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 469bbee41f13f0af4ee4fb980bbf5053@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:10:42.830: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK3119996f
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as4617a5bd
To: <sip:192.168.51.14>;tag=44DC359-198D
Date: Thu, 28 May 2015 05:10:42 GMT
Call-ID: 469bbee41f13f0af4ee4fb980bbf5053@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 5376 3244 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:10:42.838: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK46191f75
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as71c3aa9a
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 16f6c02b5725fb6f4d21f090589acc29@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:10:42.850: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK46191f75
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as71c3aa9a
To: <sip:192.168.51.14>;tag=44DC369-309
Date: Thu, 28 May 2015 05:10:42 GMT
Call-ID: 16f6c02b5725fb6f4d21f090589acc29@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 4206 4484 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:10:52.831: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK7c9524db
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as61f3a380
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 2f6ec14863e4b11956b7189a0cc2e657@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:10:52.843: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK7c9524db
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as61f3a380
To: <sip:192.168.51.14>;tag=44DEA71-408
Date: Thu, 28 May 2015 05:10:52 GMT
Call-ID: 2f6ec14863e4b11956b7189a0cc2e657@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 1875 6905 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:10:52.847: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK35dde05c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as249889b3
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 49c97ef277804a4b5a93a7132927d8c9@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:10:52.859: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK35dde05c
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as249889b3
To: <sip:192.168.51.14>;tag=44DEA85-195D
Date: Thu, 28 May 2015 05:10:52 GMT
Call-ID: 49c97ef277804a4b5a93a7132927d8c9@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 2732 4208 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:11:02.724: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK31cc6205
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as6fbf5134
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 44c8eafd5c5d99c27f26fad4672c20bd@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:11:02.736: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK31cc6205
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as6fbf5134
To: <sip:192.168.51.14>;tag=44E1115-32E
Date: Thu, 28 May 2015 05:11:02 GMT
Call-ID: 44c8eafd5c5d99c27f26fad4672c20bd@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 6231 8593 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:11:02.736: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK1bd8f995
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as55c98d3a
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 0b3116400e84aabd6ac3f8d270cf94cb@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:11:02.748: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK1bd8f995
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as55c98d3a
To: <sip:192.168.51.14>;tag=44E1125-121E
Date: Thu, 28 May 2015 05:11:02 GMT
Call-ID: 0b3116400e84aabd6ac3f8d270cf94cb@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 3918 8174 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:11:10.749: Sent:
INVITE sip:551@192.168.51.13:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.51.14:5060
From: <sip:192.168.51.14>;tag=44E3062-1F25
To: <sip:551@192.168.51.13>
Date: Thu, 28 May 2015 05:11:10 GMT
Call-ID: C9EC2C43-42E11E5-B91097EE-60ED9EEB@192.168.51.14
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 3387578010-70128101-3104675822-1626185451
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: <sip:192.168.51.14>;party=calling;screen=no;privacy=off
Timestamp: 1432789870
Contact: <sip:192.168.51.14:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 276
v=0
o=CiscoSystemsSIP-GW-UserAgent 4258 104 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 16916 RTP/AVP 18 8 0 19
c=IN IP4 192.168.51.14
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 28 05:11:10.753: Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.51.14:5060;received=192.168.51.14;rport=57249
From: <sip:192.168.51.14>;tag=44E3062-1F25
To: <sip:551@192.168.51.13>;tag=as72c9de2f
Call-ID: C9EC2C43-42E11E5-B91097EE-60ED9EEB@192.168.51.14
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.32.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03de257b"
Content-Length: 0
May 28 05:11:10.761: //91/C9EA569AB90D/CCAPI/cc_api_call_disconnect_done: cause=57,retry=0,vcCauseCode=0
May 28 05:11:10.761: //91/xxxxxxxxxxxx/CCAPI/cc_get_call_entry: callID (0x5B) not found
May 28 05:11:10.765: Sent:
ACK sip:551@192.168.51.13:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.51.14:5060
From: <sip:192.168.51.14>;tag=44E3062-1F25
To: <sip:551@192.168.51.13>;tag=as72c9de2f
Date: Thu, 28 May 2015 05:11:10 GMT
Call-ID: C9EC2C43-42E11E5-B91097EE-60ED9EEB@192.168.51.14
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
May 28 05:11:11.558: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK7e20cb53
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as3204546c
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 52335d5d66ea214b62e80a2027227c12@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:11:11.570: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK7e20cb53
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as3204546c
To: <sip:192.168.51.14>;tag=44E339C-1C6C
Date: Thu, 28 May 2015 05:11:11 GMT
Call-ID: 52335d5d66ea214b62e80a2027227c12@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 2624 2467 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:11:11.586: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK1dad2f07
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as40a55241
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 3cfaa8ca440d53863871463d248cb5e2@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:11:11.598: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK1dad2f07
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as40a55241
To: <sip:192.168.51.14>;tag=44E33B8-173
Date: Thu, 28 May 2015 05:11:11 GMT
Call-ID: 3cfaa8ca440d53863871463d248cb5e2@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 1713 6184 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:11:21.571: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK2b15132b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as6e282cf3
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 0a03190f29bae564487c13827dff0ede@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:11:21.583: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK2b15132b
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as6e282cf3
To: <sip:192.168.51.14>;tag=44E5AB4-16B
Date: Thu, 28 May 2015 05:11:21 GMT
Call-ID: 0a03190f29bae564487c13827dff0ede@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 168
v=0
o=CiscoSystemsSIP-GW-UserAgent 8522 1715 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
May 28 05:11:21.595: Received:
OPTIONS sip:192.168.51.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK52fff0d0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as34b20aac
To: <sip:192.168.51.14>
Contact: <sip:asterisk@192.168.51.13:5060>
Call-ID: 12cbb31a138f3e1c6becdc336f5b1b70@192.168.51.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.32.2
Date: Thu, 28 May 2015 05:11:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

May 28 05:11:21.607: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.51.13:5060;branch=z9hG4bK52fff0d0
From: "asterisk" <sip:asterisk@192.168.51.13>;tag=as34b20aac
To: <sip:192.168.51.14>;tag=44E5AD0-1DBC
Date: Thu, 28 May 2015 05:11:21 GMT
Call-ID: 12cbb31a138f3e1c6becdc336f5b1b70@192.168.51.13:5060
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
CSeq: 102 OPTIONS
Supported: 100rel
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 167
v=0
o=CiscoSystemsSIP-GW-UserAgent 5320 872 IN IP4 192.168.51.14
s=SIP Call
c=IN IP4 192.168.51.14
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 192.168.51.14
no debug ccsip messages
SIP Call messages tracing is disabled
cisco-2fxo#
----------

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO,

lockdoc, 08:23 , 28-Май-15 (3)

Вот вывод отладки по "debug voip rtp" во время звонка
-----------
cisco-2fxo#debug voip rtp
VOIP RTP Debug - ALL debugging is on
May 28 05:18:46.544: voip_rtp_create_session: callID=93, dstCallID=-1 laddr=192.168.51.14, lport=16938,raddr=0.0.0.0, rport=0, type=2, sig_tos=3, ip_tos=5
May 28 05:18:46.544:  voip_rtp_update_xmit_info
May 28 05:18:46.544:  voip_rtp_update_xmit_info, dstvdbptr: 0, dstCallID -1, gccb: 81DEF530, xmitFunc 0,context 0
May 28 05:18:46.544:  voip_rtp_update_xmit_info Context is NULL, exit
May 28 05:18:46.544: voip_rtp_set_non_rtp_call: Non-RTP call end
May 28 05:18:46.552: voip_rtp_stop_disc_timer
May 28 05:18:46.552: voip_rtcp_remove_ccb
May 28 05:18:46.556: //93/D997C96BB96F/CCAPI/cc_api_call_disconnect_done: cause=57,retry=0,vcCauseCode=0
May 28 05:18:46.556: voip_rtcp_close_session
May 28 05:18:46.556: voip_rtcp_stop_session:
May 28 05:18:46.556: //93/xxxxxxxxxxxx/CCAPI/cc_get_call_entry: callID (0x5D) not found
------------

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO,

users, 09:06 , 28-Май-15 (4)

Бегло глянул дебаг, вот это смутило:
> May 28 05:11:10.753: Received:
> SIP/2.0 401 Unauthorized
- в связке Циска-Астерикс как-то организован вопрос авторизации на сип-сервере?
В Астериске не силён.
С Авайским SES'ом Циска 1760 работает корректно.

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO,

lockdoc, 09:38 , 28-Май-15 (5)

> Бегло глянул дебаг, вот это смутило:
>> May 28 05:11:10.753: Received:
>> SIP/2.0 401 Unauthorized
> - в связке Циска-Астерикс как-то организован вопрос авторизации на сип-сервере?
> В Астериске не силён.
> С Авайским SES'ом Циска 1760 работает корректно.
Вот так:
---------
[703]
type=friend
host=192.168.51.14
qualify=yes
qualifyfreq=10
disallow=all
allow=ulaw
allow=alaw
context=main
nat=no
insecury=invite
--------
Здесь что то не так?

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO,

ShyLion, 11:18 , 28-Май-15 (6)


May 28 05:11:10.749: Sent:
INVITE sip:551@192.168.51.13:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.51.14:5060
From: <sip:192.168.51.14>;tag=44E3062-1F25
To: <sip:551@192.168.51.13>
Date: Thu, 28 May 2015 05:11:10 GMT


May 28 05:11:10.753: Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.51.14:5060;received=192.168.51.14;rport=57249
From: <sip:192.168.51.14>;tag=44E3062-1F25
To: <sip:551@192.168.51.13>;tag=as72c9de2f
Call-ID: C9EC2C43-42E11E5-B91097EE-60ED9EEB@192.168.51.14
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.32.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03de257b"
Content-Length: 0

Астериск "друга" не узнает и требует логин-пароль.

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO,

ShyLion, 11:21 , 28-Май-15 (7)

Вообще по моему киса на астериске должна быть peer в твоей ситуации. А номера 551 и т.д. они же на других аппаратах живут а не на кисе, она лишь посредник.

Ну и дергать ее каждые 10 секунд нет смысла, если нет альтернативных маршрутов.

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO, lockdoc, 13:44 , 28-Май-15 (8)
> Вообще по моему киса на астериске должна быть peer в твоей ситуации.
> А номера 551 и т.д. они же на других аппаратах
> живут а не на кисе, она лишь посредник.
Думаю это не важно, поменял на peer, ничего не изменилось, таже ошибка, астериск не пускает киску для звонка, хотя должен, так как астериску явно указано в директире:
host:192.168.51.14
> Ну и дергать ее каждые 10 секунд нет смысла, если нет альтернативных
> маршрутов.
Здесь строку с дерганием можно убрать, я не против. Поставил ее для быстрого обнаружения отвала киски.
Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO, lockdoc, 13:46 , 28-Май-15 (9)
>[оверквотинг удален]
> To: <sip:551@192.168.51.13>;tag=as72c9de2f
> Call-ID: C9EC2C43-42E11E5-B91097EE-60ED9EEB@192.168.51.14
> CSeq: 101 INVITE
> Server: Asterisk PBX 1.8.32.2
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03de257b"
> Content-Length: 0
>
> Астериск "друга" не узнает и требует логин-пароль.
Хотя должна...
Можен у меня криво астериск установился?

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO,

lockdoc, 13:55 , 28-Май-15 (10)

С астриском думаю все нормально, проверял данную киску на другом рабочем сервере астериска результат - тот же. Вот лог
-------
May 28 10:49:58.687: Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.51.14:5060;received=84.53.241.XXX;rport=56601
From: <sip:192.168.51.14>;tag=5845B46-1929
To: <sip:551@84.53.245.XX:5060>;tag=as6b1cbe6d
Call-ID: 1E4B512E-45E11E5-BC9397EE-60ED9EEB@192.168.51.14
CSeq: 101 INVITE
Server: XXXXXXXXXXXXXXXXXXXXXX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7164db16"
Content-Length: 0

May 28 10:49:58.695: //137/1E4A1846BC90/CCAPI/cc_api_call_disconnect_done: cause=57,retry=0,vcCauseCode=0
May 28 10:49:58.695: //137/xxxxxxxxxxxx/CCAPI/cc_get_call_entry: callID (0x89) not found
May 28 10:49:58.699: Sent:
ACK sip:551@84.53.245.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.51.14:5060
From: <sip:192.168.51.14>;tag=5845B46-1929
To: <sip:551@84.53.245.XX>;tag=as6b1cbe6d
Date: Thu, 28 May 2015 10:49:58 GMT
Call-ID: 1E4B512E-45E11E5-BC9397EE-60ED9EEB@192.168.51.14
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
-------------------

Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO, lockdoc, 14:06 , 28-Май-15 (11)
В настройках астериска поставил
allowguest=yes
Результат такой же :-(
Вот лог киски:
----May 28 11:03:31.762: Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.51.14:5060;received=192.168.51.14;rport=54362
From: <sip:192.168.51.14>;tag=590C366-EE1
To: <sip:551@192.168.51.13>
Call-ID: 2F22D5A-46011E5-BCAC97EE-60ED9EEB@192.168.51.14
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.32.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:551@192.168.51.13:5060>
Content-Length: 0
May 28 11:03:31.766: Received:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.51.14:5060;received=192.168.51.14;rport=54362
From: <sip:192.168.51.14>;tag=590C366-EE1
To: <sip:551@192.168.51.13>;tag=as36de75c4
Call-ID: 2F22D5A-46011E5-BCAC97EE-60ED9EEB@192.168.51.14
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.32.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0
May 28 11:03:31.774: //145/02F0F3D9BCA9/CCAPI/cc_api_call_disconnect_done: cause=21,retry=0,vcCauseCode=0
May 28 11:03:31.774: //145/xxxxxxxxxxxx/CCAPI/cc_get_call_entry: callID (0x91) not found
May 28 11:03:31.778: Sent:
ACK sip:551@192.168.51.13:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.51.14:5060
From: <sip:192.168.51.14>;tag=590C366-EE1
To: <sip:551@192.168.51.13>;tag=as36de75c4
Date: Thu, 28 May 2015 11:03:31 GMT
Call-ID: 2F22D5A-46011E5-BCAC97EE-60ED9EEB@192.168.51.14
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
--------
При том, что ошибка уже 603
(SIP/2.0 603 Decline - вызываемый пользователь не желает принимать входящие вызовы, не указывая причину отказа)
Вот  блин!
Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO, lockdoc, 23:13 , 28-Май-15 (12)
Вроде разобрался.....
Если ставить allowguest=yes то звонок проходит, но попадает на дефолтовый (default) номерной план (????). В астериске явно указано, что
-------
[703]
type=peer
host=192.168.51.14
qualify=yes
qualifyfreq=200
dtfmmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
context=main
nat=no
insecury=port,invite
----------
Хост 192.168.51.14 это 703, но астериск его не определяет почему то????
Вот лог
-------
== Using SIP RTP CoS mark 5
    -- Executing [551@default:1] Answer("SIP/192.168.51.14-00000000", "") in new stack
    -- Executing [551@default:2] Playback("SIP/192.168.51.14-00000000", "beep") in new stack
    -- <SIP/192.168.51.14-00000000> Playing 'beep.ulaw' (language 'en')
    -- Executing [551@default:3] SayUnixTime("SIP/192.168.51.14-00000000", ",,QAdBk'digits/oclock'M'minutes'") in new stack
    -- <SIP/192.168.51.14-00000000> Playing 'digits/today.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/day-4.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/20.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/h-8.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/mon-4.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/20.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/2.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/oclock.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/40.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'digits/5.ulaw' (language 'en')
    -- <SIP/192.168.51.14-00000000> Playing 'minutes.ulaw' (language 'en')
    -- Executing [551@default:4] Dial("SIP/192.168.51.14-00000000", "SIP/119,60,t") in new stack
  == Using SIP RTP CoS mark 5
------------
Происходит входящий анонимный звонок, а когда они запрещены allowguest=no, то отбой.
Теперь ломаю голову на правильным описанием пира киски...
Может у кого то есть какие-нибудь мысли по этому вопросу?
Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO, lockdoc, 23:32 , 28-Май-15 (13)
>[оверквотинг удален]
> host=192.168.51.14
> qualify=yes
> qualifyfreq=200
> dtfmmode=rfc2833
> disallow=all
> allow=ulaw
> allow=alaw
> context=main
> nat=no
> insecury=port,invite
А вот и ошибка, черт побери....................
insecure=port,invite
>[оверквотинг удален]
>     -- <SIP/192.168.51.14-00000000> Playing 'digits/40.ulaw' (language 'en')
>     -- <SIP/192.168.51.14-00000000> Playing 'digits/5.ulaw' (language 'en')
>     -- <SIP/192.168.51.14-00000000> Playing 'minutes.ulaw' (language 'en')
>     -- Executing [551@default:4] Dial("SIP/192.168.51.14-00000000", "SIP/119,60,t") in
> new stack
>   == Using SIP RTP CoS mark 5
> ------------
> Происходит входящий анонимный звонок, а когда они запрещены allowguest=no, то отбой.
> Теперь ломаю голову на правильным описанием пира киски...
> Может у кого то есть какие-нибудь мысли по этому вопросу?
Ура все заработало!!!!
Три дня выноса мозга... Зато теперь в астериске стал лучше разбираться!!
Ответить | Сообщить модератору

Нужна помощь в настройке Cisco 1760 c 2FXO, ShyLion, 07:14 , 29-Май-15 (14)
Молодец
Ответить | Сообщить модератору