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спросил 2013-10-30 10:01:32 +0400

vdvas Gravatar vdvas

не правильно определяется номер

Здравствуйте. asterisk 1.8 + cisco spa 8000. Появилась проблема при звонке по внутренним номерам - неправильно определяется номер. Внутренняя адресация 7XX, при звонке с 701 номера определяется как 801. Думал что проблема в номерах для парковки вызовов (конфликт), сменил номера для парковки - проблема осталась. Что можно ещё посмотреть?

не правильно определяется номер

Здравствуйте. asterisk 1.8 + cisco spa 8000. Появилась проблема при звонке по внутренним номерам - неправильно определяется номер. Внутренняя адресация 7XX, при звонке с 701 номера определяется как 801. Думал что проблема в номерах для парковки вызовов (конфликт), сменил номера для парковки - проблема осталась. Что можно ещё посмотреть?

Звонок с 750 на 701. запустил дебаг на пире 701.

 Audio is at 12078
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 Reliably Transmitting (no NAT) to 192.168.1.82:5060:
 INVITE sip:701@192.168.1.82:5060 SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
 Max-Forwards: 70
 From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
 To: <sip:701@192.168.1.82:5060>
 Contact: <sip:750@192.168.1.73:5060>
 Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.8.23.1
 Date: Thu, 31 Oct 2013 08:17:54 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 264

 v=0
 o=root 1357782223 1357782223 IN IP4 192.168.1.73
 s=Asterisk PBX 1.8.23.1
 c=IN IP4 192.168.1.73
 t=0 0
 m=audio 12078 RTP/AVP 8 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv

 ---

 <--- SIP read from UDP:192.168.1.82:5060 --->
 SIP/2.0 100 Trying
 To: <sip:701@192.168.1.82:5060>
 From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
 Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
 CSeq: 102 INVITE
 Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
 Server: Linksys/SPA8000-6.1.12(XU)
 Allow-Events: talk, hold, conference
 Content-Length: 0

 <------------->
  --- (9 headers 0 lines) ---

 <--- SIP read from UDP:192.168.1.82:5060 --->
 SIP/2.0 180 Ringing
 To: <sip:701@192.168.1.82:5060>;tag=8ca32beb6c4a5562i0
 From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
 Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
 CSeq: 102 INVITE
 Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
 Contact: "701" <sip:701@192.168.1.82:5060>
 Server: Linksys/SPA8000-6.1.12(XU)
 Remote-Party-ID: "701" <sip:701@192.168.1.73>;screen=yes;party=called
 Allow-Events: talk, hold, conference
 Content-Length: 0

  <------------->
 --- (11 headers 0 lines) ---
 list_route: hop: <sip:701@192.168.1.82:5060>

 <--- SIP read from UDP:192.168.1.82:5060 --->
  SIP/2.0 480 Temporarily not available
 To: <sip:701@192.168.1.82:5060>;tag=8ca32beb6c4a5562i0
  From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
 Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
  CSeq: 102 INVITE
  Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
  Server: Linksys/SPA8000-6.1.12(XU)
  Allow-Events: talk, hold, conference
  Content-Length: 0

  <------------->
  --- (9 headers 0 lines) ---
   set_destination: Parsing <sip:701@192.168.1.82:5060> for address/port to send to
  set_destination: set destination to 192.168.1.82:5060
  Transmitting (no NAT) to 192.168.1.82:5060:
  ACK sip:701@192.168.1.82:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
  Max-Forwards: 70
  From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
  To: <sip:701@192.168.1.82:5060>;tag=8ca32beb6c4a5562i0
  Contact: <sip:750@192.168.1.73:5060>
  Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
  CSeq: 102 ACK 
  User-Agent: Asterisk PBX 1.8.23.1
  Content-Length: 0


   ---
   Really destroying SIP dialog '5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060' Method: INVITE

не правильно определяется номер

Здравствуйте. asterisk 1.8 + cisco spa 8000. Появилась проблема при звонке по внутренним номерам - неправильно определяется номер. Внутренняя адресация 7XX, при звонке с 701 номера определяется как 801. Думал что проблема в номерах для парковки вызовов (конфликт), сменил номера для парковки - проблема осталась. Что можно ещё посмотреть?

Звонок с 750 на 701. запустил дебаг на пире 701.

 Audio is at 12078
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 Reliably Transmitting (no NAT) to 192.168.1.82:5060:
 INVITE sip:701@192.168.1.82:5060 SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
 Max-Forwards: 70
 From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
 To: <sip:701@192.168.1.82:5060>
 Contact: <sip:750@192.168.1.73:5060>
 Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.8.23.1
 Date: Thu, 31 Oct 2013 08:17:54 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 264

 v=0
 o=root 1357782223 1357782223 IN IP4 192.168.1.73
 s=Asterisk PBX 1.8.23.1
 c=IN IP4 192.168.1.73
 t=0 0
 m=audio 12078 RTP/AVP 8 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv

 ---

 <--- SIP read from UDP:192.168.1.82:5060 --->
 SIP/2.0 100 Trying
 To: <sip:701@192.168.1.82:5060>
 From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
 Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
 CSeq: 102 INVITE
 Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
 Server: Linksys/SPA8000-6.1.12(XU)
 Allow-Events: talk, hold, conference
 Content-Length: 0

 <------------->
  --- (9 headers 0 lines) ---

 <--- SIP read from UDP:192.168.1.82:5060 --->
 SIP/2.0 180 Ringing
 To: <sip:701@192.168.1.82:5060>;tag=8ca32beb6c4a5562i0
 From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
 Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
 CSeq: 102 INVITE
 Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
 Contact: "701" <sip:701@192.168.1.82:5060>
 Server: Linksys/SPA8000-6.1.12(XU)
 Remote-Party-ID: "701" <sip:701@192.168.1.73>;screen=yes;party=called
 Allow-Events: talk, hold, conference
 Content-Length: 0

  <------------->
 --- (11 headers 0 lines) ---
 list_route: hop: <sip:701@192.168.1.82:5060>

 <--- SIP read from UDP:192.168.1.82:5060 --->
  SIP/2.0 480 Temporarily not available
 To: <sip:701@192.168.1.82:5060>;tag=8ca32beb6c4a5562i0
  From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
 Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
  CSeq: 102 INVITE
  Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
  Server: Linksys/SPA8000-6.1.12(XU)
  Allow-Events: talk, hold, conference
  Content-Length: 0

  <------------->
  --- (9 headers 0 lines) ---
   set_destination: Parsing <sip:701@192.168.1.82:5060> for address/port to send to
  set_destination: set destination to 192.168.1.82:5060
  Transmitting (no NAT) to 192.168.1.82:5060:
  ACK sip:701@192.168.1.82:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK7e311b6e
  Max-Forwards: 70
  From: "750" <sip:750@192.168.1.73>;tag=as461ab9af
  To: <sip:701@192.168.1.82:5060>;tag=8ca32beb6c4a5562i0
  Contact: <sip:750@192.168.1.73:5060>
  Call-ID: 5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060
  CSeq: 102 ACK 
  User-Agent: Asterisk PBX 1.8.23.1
  Content-Length: 0


   ---
   Really destroying SIP dialog '5fefcbd30071183300b1d0d726f81d47@192.168.1.73:5060' Method: INVITE

Звонок с 750 на 701. core set verbose 3. sip set debug peer 701.

-- Executing [701@yar-out:1] Dial("SIP/750-000005ab", "SIP/701,,Tt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 17416
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.82:5060:
INVITE sip:701@192.168.1.82:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK362227f9
Max-Forwards: 70
From: "750" <sip:750@192.168.1.73>;tag=as16c4c791
To: <sip:701@192.168.1.82:5060>
Contact: <sip:750@192.168.1.73:5060>
Call-ID: 78047d6a46b4f9d61138fec150940368@192.168.1.73:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.1
Date: Thu, 31 Oct 2013 10:04:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1907214390 1907214390 IN IP4 192.168.1.73
s=Asterisk PBX 1.8.23.1
c=IN IP4 192.168.1.73
t=0 0
m=audio 17416 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called SIP/701

<--- SIP read from UDP:192.168.1.82:5060 --->
SIP/2.0 100 Trying
To: <sip:701@192.168.1.82:5060>
From: "750" <sip:750@192.168.1.73>;tag=as16c4c791
Call-ID: 78047d6a46b4f9d61138fec150940368@192.168.1.73:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK362227f9
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.82:5060 --->
SIP/2.0 180 Ringing
To: <sip:701@192.168.1.82:5060>;tag=e31d60c16470841ai0
From: "750" <sip:750@192.168.1.73>;tag=as16c4c791
Call-ID: 78047d6a46b4f9d61138fec150940368@192.168.1.73:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK362227f9
Contact: "701" <sip:701@192.168.1.82:5060>
Server: Linksys/SPA8000-6.1.12(XU)
Remote-Party-ID: "701" <sip:701@192.168.1.73>;screen=yes;party=called
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:701@192.168.1.82:5060>
 -- SIP/701-000005ac is ringing
Scheduling destruction of SIP dialog         '78047d6a46b4f9d61138fec150940368@192.168.1.73:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.1.82:5060:
CANCEL sip:701@192.168.1.82:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK362227f9
Max-Forwards: 70
From: "750" <sip:750@192.168.1.73>;tag=as16c4c791
To: <sip:701@192.168.1.82:5060>
Call-ID: 78047d6a46b4f9d61138fec150940368@192.168.1.73:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.23.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '78047d6a46b4f9d61138fec150940368@192.168.1.73:5060' in 32000 ms (Method: INVITE)
== Spawn extension (yar-out, 701, 1) exited non-zero on 'SIP/750-000005ab'
-- Executing [h@yar-out:1] Hangup("SIP/750-000005ab", "") in new stack
 == Spawn extension (yar-out, h, 1) exited non-zero on 'SIP/750-000005ab'

<--- SIP read from UDP:192.168.1.82:5060 --->
 SIP/2.0 487 Request Terminated
 To: <sip:701@192.168.1.82:5060>;tag=e31d60c16470841ai0
From: "750" <sip:750@192.168.1.73>;tag=as16c4c791
Call-ID: 78047d6a46b4f9d61138fec150940368@192.168.1.73:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK362227f9
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.82:5060:
ACK sip:701@192.168.1.82:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK362227f9
Max-Forwards: 70
From: "750" <sip:750@192.168.1.73>;tag=as16c4c791
To: <sip:701@192.168.1.82:5060>;tag=e31d60c16470841ai0
Contact: <sip:750@192.168.1.73:5060>
Call-ID: 78047d6a46b4f9d61138fec150940368@192.168.1.73:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.1
Content-Length: 0


---
Scheduling destruction of SIP dialog    '78047d6a46b4f9d61138fec150940368@192.168.1.73:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.82:5060 --->
SIP/2.0 200 OK
To: <sip:701@192.168.1.82:5060>;tag=e31d60c16470841ai0
From: "750" <sip:750@192.168.1.73>;tag=as16c4c791
Call-ID: 78047d6a46b4f9d61138fec150940368@192.168.1.73:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.1.73:5060;branch=z9hG4bK362227f9
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 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.