esveka » 04 окт 2010, 22:00
Шлю лучи счастья этому форуму.
Помогите разобраться. Подключил мультифон. Есть входящие - слышно в обе стороны. Есть исходящие - слышо только меня. Клиент cisco ata 186 . Кодеками поигрался. Привожу sip debug
Я....прошу...извинить. Хотел присоединить файл с дебагом, но что-то не вижу его. Поэтому - в тело.
*CLI>
<--- SIP read from UDP:192.168.1.5:5060 --->
BYE sip:7987XXXXXXX@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060
From: <sip:2005@192.168.1.2;user=phone>;tag=990052330
To: <sip:7987XXXXXXX@192.168.1.2;user=phone>;tag=as49933b61
Call-ID: 964782398@192.168.1.5
CSeq: 3 BYE
User-Agent: Cisco ATA 186 v3.1.1 atasip (040629A)
Authorization: Digest username="2005",realm="asterisk",nonce="3e98fa40",uri="sip:7987XXXXXXX@192.168.1.2",response="4949b674f7d2bcadec994dd9deb7161c"
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.1.5 : 5060 (NAT)
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;received=192.168.1.5
From: <sip:2005@192.168.1.2;user=phone>;tag=990052330
To: <sip:7987XXXXXXX@192.168.1.2;user=phone>;tag=as49933b61
Call-ID: 964782398@192.168.1.5
CSeq: 3 BYE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (incoming, 7987XXXXXXX, 1) exited non-zero on 'SIP/2005-00000012'
Really destroying SIP dialog '964782398@192.168.1.5' Method: BYE
Reliably Transmitting (NAT) to 192.168.1.5:5060:
OPTIONS sip:2005@192.168.1.5:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3aa64262;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as7598f6e5
To: <sip:2005@192.168.1.5:5060;user=phone;transport=udp>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 12a3b45f4b28243348a6154e53607c36@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Mon, 04 Oct 2010 17:42:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3aa64262;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as7598f6e5
To: <sip:2005@192.168.1.5:5060;user=phone;transport=udp>;tag=990052330
Call-ID: 12a3b45f4b28243348a6154e53607c36@192.168.1.2
CSeq: 102 OPTIONS
Server: Cisco ATA 186 v3.1.1 atasip (040629A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Supported: replaces
Content-Length: 268
Content-Type: application/sdp
v=0
o=2005 29183 29183 IN IP4 192.168.1.5
s=ATA186 Call
c=IN IP4 192.168.1.5
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 12 lines) ---
Really destroying SIP dialog '12a3b45f4b28243348a6154e53607c36@192.168.1.2' Method: OPTIONS