asterisk.ru

Код: Выделить всё

Got SIP response 603 "Declined" back from "voip_шлюз":5060
    -- SIP/912-00000062 is busy

Код: Выделить всё

== Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x40 (slin) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to "fxo_шлюз":
INVITE sip:89180494402@"fxo_шлюз" SIP/2.0
Via: SIP/2.0/UDP "ip_atc";branch=z9hG4bK321c06c3
Max-Forwards: 70
From: "Kudarev NN" <sip:"username"@"ip_atc">;tag=as19530cb8
To: <sip:89180494402@"fxo_шлюз">
Contact: <sip:"username"@"ip_atc">
Call-ID: 7e7f873f494e657a2d4457df785752f2@"ip_atc"
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.7.1)
Date: Thu, 03 Sep 2015 11:31:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 386

v=0
o=root 489176493 489176493 IN IP4 "ip_atc"
s=Asterisk PBX 1.8.7.1
c=IN IP4 "ip_atc"
t=0 0
m=audio 14150 RTP/AVP 0 8 3 18 10 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:10 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/912/89180494402

<--- SIP read from UDP:"fxo_шлюз" --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP "ip_atc":5060;branch=z9hG4bK321c06c3;received="ip_atc"
From: "Kudarev NN" <sip:"username"@"ip_atc">;tag=as19530cb8
To: <sip:89180494402@"fxo_шлюз">
Call-ID: 7e7f873f494e657a2d4457df785752f2@"ip_atc"
CSeq: 102 INVITE
Server: TA810
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89180494402@"fxo_шлюз">
Content-Length: 0

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

<--- SIP read from UDP:"fxo_шлюз"--->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP "ip_atc";branch=z9hG4bK321c06c3;received="ip_atc"
From: "Kudarev NN" <sip:223@"ip_atc">;tag=as19530cb8
To: <sip:89180494402@"fxo_шлюз">;tag=as6a72c19d
Call-ID: 7e7f873f494e657a2d4457df785752f2@"ip_atc"
CSeq: 102 INVITE
Server: TA810
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Got SIP response 603 "Declined" back from "fxo_шлюз":5060
Transmitting (no NAT) to "fxo_шлюз":5060:
ACK sip:89180494402@"fxo_шлюз" SIP/2.0
Via: SIP/2.0/UDP "ip_atc":5060;branch=z9hG4bK321c06c3
Max-Forwards: 70
From: "Kudarev NN" <sip:"username"@"ip_atc">;tag=as19530cb8
To: <sip:XXXXXXXXXXX@"fxo_шлюз">;tag=as6a72c19d
Contact: <sip:"username"@"ip_atc">
Call-ID: 7e7f873f494e657a2d4457df785752f2@"ip_atc"
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(1.8.7.1)
Content-Length: 0


---
    -- SIP/912-00000083 is busy

Код: Выделить всё

Got SIP response 603 "Declined" back from "fxo_шлюз" 
 -- SIP/912-00000083 is busy