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Отбой входящего звонка при поднятии трубки

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Добрый день. Есть Asterisk + FreePBX

Есть удалённый клиент на SIP-шлюзе Grandstream. При поступлении на этот сип звонка, сам звонок проходит, но при снятии трубки происходит отбой. Исходящие звонки с этого клиента осуществляются нормально.

В дебаге пишет такое:

    <--- SIP read from UDP:83.83.216.181:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK376e33ab;rport=5060
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 102 INVITE
Contact: <sip:103@83.83.216.181:5061>
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.2A 1.0.1.41
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 2450
o=103 8002 8000 IN IP4 83.83.216.181
s=SIP Call
c=IN IP4 83.83.216.181
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
<------------->
--- (23 headers 0 lines) ---
[2015-11-16 22:27:18] WARNING[1800][C-000039c0]: chan_sip.c:23033 handle_response_invite: Received response: "200 OK" from '103' without SDP
list_route: hop: <sip:103@83.83.216.181:5061>
set_destination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to
set_destination: set destination to 83.83.216.181:5061
Transmitting (NAT) to 83.83.216.181:5061:
ACK sip:103@83.83.216.181:5061 SIP/2.0
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK7a852fc1;rport
Max-Forwards: 70
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Contact: <sip:500@109.74.139.75:5060>
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
Content-Length: 0


---
set_destination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to
set_destination: set destination to 83.83.216.181:5061
Reliably Transmitting (NAT) to 83.83.216.181:5061:
BYE sip:103@83.83.216.181:5061 SIP/2.0
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK08fe6c29;rport
Max-Forwards: 70
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 103 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
Scheduling destruction of SIP dialog '0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to
set_destination: set destination to 83.83.216.181:5061
Reliably Transmitting (NAT) to 83.83.216.181:5061:
BYE sip:103@83.83.216.181:5061 SIP/2.0
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK23edd013;rport
Max-Forwards: 70
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 104 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:83.83.216.181:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK08fe6c29;rport=5060
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 103 BYE
Contact: <sip:103@83.83.216.181:5061>
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.2A 1.0.1.41
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

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

<--- SIP read from UDP:83.83.216.181:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK23edd013;rport=5060
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 104 BYE
Contact: <sip:103@83.83.216.181:5061>
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.2A 1.0.1.41
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060' Method: INVITE
удалить закрыть спам изменить тег редактировать

спросил 2015-11-17 00:06:47 +0400

georgys Gravatar georgys
1 1

обновил 2015-11-17 01:56:40 +0400

zzuz Gravatar zzuz flag of Russian Federation
6744 2 6 69
http://line24.ru/

Comments

sip*CLI> sip show peer 103

  • Name : 103 Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : MoscowAndInternal Record On feature : automon Record Off feature : automon Subscr.Cont. : <Not set> Language : ru Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "103" <103> MaxCallBR : 384 kbps Expire : 778 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No
georgys ( 2015-11-17 00:12:38 +0400 )редактировать

Ключевая фраза "[2015-11-16 22:27:18] WARNING[1800][C-000039c0]: chansip.c:23033 handleresponse_invite: Received response: "200 OK" from '103' without SDP "

zzuz ( 2015-11-17 01:57:57 +0400 )редактировать

А что делать с этим without SDP??? Я уже весь гугл скурил ((((

georgys ( 2015-11-17 16:37:45 +0400 )редактировать

Обновить прошивку аппарата или настроить его .

zzuz ( 2015-11-17 16:55:31 +0400 )редактировать

Будьте первым, кто ответит на этот вопрос!

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Задан: 2015-11-17 00:06:47 +0400

Просмотрен: 228 раз

Обновлен: Nov 17 '15

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