asterisk =>SIP trunk => cisco callmanager 4.1 => cisco 2821 => SIP => VPS
trunk зарегистрирован на asterisk, звонки в город уходят нормально, а вот на мобильные - трабла.
extensions.conf
Код: Выделить всё
[general]
static=yes
writeprotect=no
clearglobalvars=no
context=internal
; CALLS TO LOCAL ASTERISK
[internal]
exten => _XXXX,1,Dial(SIP/${EXTEN})
exten => _XXXX,n,Hangup
include => to_callman01
; CALLS TO CISCO CALLMANAGER
[to_callman01]
exten => _XXX,1,Dial(SIP/callman01/${EXTEN})
exten => _XXX,n,Hangup
exten => _08495XXXXXXX,1,Dial(SIP/callman01/${EXTEN}) - звонок в город проходит
exten => _089XXXXXXXXX,1,Dial(SIP/callman01/${EXTEN}) - звонок на мобильные ошибка
include => internalКод: Выделить всё
[callman01]
type=friend
context=to_callman01
host=192.168.10.3
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=no
insecure=port,invite
на 2821 отшибаются 0 и оператору уходит в нужном формате.
ошибка в asterisk>
Код: Выделить всё
== Using SIP RTP CoS mark 5
-- Executing [089251234567@internal:1] Dial("SIP/1000-0000004c", "SIP/callman01/089251234567") in new stack
== Using SIP RTP CoS mark 5
Audio is at 17556
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.10.3:5060:
INVITE sip:089251234567@192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.1:5060;branch=z9hG4bK554c23f6
Max-Forwards: 70
From: "1000" <sip:1000@172.16.10.1>;tag=as6c89fbc6
To: <sip:089251234567@192.168.10.3>
Contact: <sip:1000@172.16.10.1:5060>
Call-ID: 260df8036a71ac4a76caebf24f1c12d2@172.16.10.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.3.0
Date: Tue, 09 Apr 2013 07:16:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257
v=0
o=root 1457650835 1457650835 IN IP4 172.16.10.1
s=Asterisk PBX 11.3.0
c=IN IP4 172.16.10.1
t=0 0
m=audio 17556 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/callman01/089251234567
<--- SIP read from UDP:192.168.10.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.10.1:5060;branch=z9hG4bK554c23f6
From: "1000" <sip:1000@172.16.10.1>;tag=as6c89fbc6
To: <sip:089251234567@192.168.10.3>;tag=17090459
Date: Tue, 09 Apr 2013 07:16:25 GMT
Call-ID: 260df8036a71ac4a76caebf24f1c12d2@172.16.10.1:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:192.168.10.3:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.16.10.1:5060;branch=z9hG4bK554c23f6
From: "1000" <sip:1000@172.16.10.1>;tag=as6c89fbc6
To: <sip:089251234567@192.168.10.3>;tag=17090459
Date: Tue, 09 Apr 2013 07:16:25 GMT
Call-ID: 260df8036a71ac4a76caebf24f1c12d2@172.16.10.1:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 192.168.10.3:5060:
ACK sip:089251234567@192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.1:5060;branch=z9hG4bK554c23f6
Max-Forwards: 70
From: "1000" <sip:1000@172.16.10.1>;tag=as6c89fbc6
To: <sip:089251234567@192.168.10.3>;tag=17090459
Contact: <sip:1000@172.16.10.1:5060>
Call-ID: 260df8036a71ac4a76caebf24f1c12d2@172.16.10.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.3.0
Content-Length: 0
---
Scheduling destruction of SIP dialog '260df8036a71ac4a76caebf24f1c12d2@172.16.10.1:5060' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/1000-0000004c' status is 'CHANUNAVAIL'
