SIP вызов без регистрации
задача: есть аккаунт у провайдера, нужно звонить через него с авторизацией, но без регистрации нескольким серверам. пытаюсь звонить так:
[Mar 28 12:09:24] VERBOSE[7208] logger.c: -- Executing [08927338922@from-internal:1] Dial("SIP/100-b77070e8", "SIP/933713
:112233@mera.ufacom.ru/8927338922") in new stack
[Mar 28 12:09:24] WARNING[7208] chan_sip.c: No such host: mera.ufacom.ru/8927338922
[Mar 28 12:09:24] WARNING[7208] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Mar 28 12:09:24] VERBOSE[7208] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
в сторону прова никаких SIP пакетов. пробую так:
<------------>
-- Executing [08927338922@from-internal:1] Dial("SIP/100-0864dd30", "SIP/933713:112233@mera.ufacom.ru") in new stack
Audio is at 192.168.0.5 port 11892
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.133.161.10:5060:
INVITE sip:933713:112233@mera.ufacom.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK1780a4fe;rport
From: "device" <sip:100@192.168.0.5>;tag=as4a7c6dd4
To: <sip:933713:112233@mera.ufacom.ru>
Contact: <sip:100@192.168.0.5>
Call-ID: 439ef8f83aab85200ec82e703d1c8253@192.168.0.5
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 28 Mar 2009 16:26:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 4654 4654 IN IP4 192.168.0.5
s=session
c=IN IP4 192.168.0.5
t=0 0
m=audio 11892 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 933713:112233@mera.ufacom.ru
trixbox1*CLI>
<--- SIP read from 62.133.161.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK1780a4fe;rport
From: "device" <sip:100@192.168.0.5>;tag=as4a7c6dd4
To: <sip:933713:112233@mera.ufacom.ru>;tag=ffff3500ffffff10ff00000e0c4affff
Call-ID: 439ef8f83aab85200ec82e703d1c8253@192.168.0.5
CSeq: 102 INVITE
Contact: <sip:933713@62.133.161.10;user=phone>
Server: MERA MSIP v.1.0.2
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 62.133.161.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK1780a4fe;rport
From: "device" <sip:100@192.168.0.5>;tag=as4a7c6dd4
To: <sip:933713:112233@mera.ufacom.ru>;tag=ffff3500ffffff10ff00000e0c4affff
Call-ID: 439ef8f83aab85200ec82e703d1c8253@192.168.0.5
CSeq: 102 INVITE
Server: MERA MSIP v.1.0.2
Reason: Q.850;cause=3;text="No route to destination"
Content-Length: 0
---
-- SIP/mera.ufacom.ru-086a7a10 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
варианты:
Dial(SIP/933713:112233@mera.ufacom.ru/8927338922) - не находит хост
Dial(SIP/933713:112233@mera.ufacom.ru) - авторизуется, но не находит абонента
Dial(SIP/8927338922@933713:112233@mera.ufacom.ru) - не работает
что еще можно попробовать? |
