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dial_exec_full: Unable to create channel of type 'SIP' (caus

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Ответить
terra2039
Сообщения: 7
Зарегистрирован: 24 апр 2011, 12:36

dial_exec_full: Unable to create channel of type 'SIP' (caus

Сообщение terra2039 »

при исходящем вызове получаю ошибку.

Код: Выделить всё

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
sip.conf

Код: Выделить всё

 [general]

    register = 74959845383:12345@sip.discounttelecom.ru		 
    bindport = 5060					  
    bindaddr = 0.0.0.0					  
    context=default					 
    allowguest=yes 					 
    allowoverlap=no					 
    realm=McConfig VoIP Server				 
	
    allow = g711
    allow = alaw						 
    srvlookup=yes					 
    language=ru 					 
    udpbindaddr=0.0.0.0
    tcpenable=no
    tcpbindaddr=0.0.0.0

 
    [authentication]
 
    [sipnet] 						 
    type=friend 					 
    insecure=invite					 
    secret=12345					 
    username=74959845383					 
    fromuser=74959845383 					 
    fromdomain=sip.discounttelecom.ru 				 
    host=sip.discounttelecom.ru 					 
    port=5060 						 
    dtmfmode=rfc2833 					 
    allow = g711,A-law
    allow = alaw					 
    canreinvite=no 					 
						         

;============================ Users ===========================
[1000] ;ITCrowd						 
    qualify=yes						 
							 
    context=LocalAndPSTNAndMobileAndLD			 
    type=friend						 
    username=1000					 
    secret=						 
    nat=no						 
    allow=g711						 
    host=dynamic					 
    callerid="IT Crowd"					 
    callgroup=1						  
    pickupgroup=1					 
    language=ru						 
    canreinvite=yes					 
						         
    mailbox=1000@fantasyworld.org,1000			 

[1001] ;Support
    qualify=yes
    context=LocalAndPSTNAndMobileAndLD
    type=friend
    username=1001
    secret=
    nat=yes
    allow=all
    host=dynamic
    callerid="Support"
    callgroup=1
    pickupgroup=1
    language=ru
    canreinvite=yes
    mailbox=1001@fantasyworld.org,1001
extensions.conf

Код: Выделить всё

[general]
    priorityjumping=yes				 
    static=yes 					 
    writeprotect=no				 
    autofallthrough=yes				 
    clearglobalvars=no				 

 
[globals]
    DYNAMIC_FEATURES=automon
     
[default]
    include => 74959845383	
    include => OSoftSipNet
    include => NSoftSipNet			 
    include => LocalOnly			 
    include => LocalAndPSTN
    include => LocalAndPSTNAndMobile
    include => LocalAndPSTNAndMobileAndLD

;==================== Start SIPNET IN =======================
 
[74959845383]
    exten => s,1,Answer				 
    exten => s,2,Set(TIMEOUT(digit)=5)		 
    exten => s,3,Set(TIMEOUT(response)=10)	 
    exten => s,4,BackGround(test/1)		 
    exten => s,5,WaitExten(5)			 
    exten => s,6,Dial(SIP/1000,20,otmw)		 
    exten => s,7,Voicemail			 
    exten => s,8,Hangup				 
    exten => i,1,Dial(SIP/1000,300,otmw)
    exten => i,2,Hangup
    exten => _97XXXX,1,Goto(LocalAndPSTNAndMobileAndLD,${EXTEN},1)
;================== end SIPNET IN ==========================
;================== Start CONTEXT ==========================
  
[OSoftSipNet]
    exten => _9XXXXXXX,1,Dial(SIP/${EXTEN:1}@sip.discounttelecom.ru)

 
[NSoftSipNet]
    include => OSoftSipNet
    exten => _900XXXXXXXX,1,Dial(SIP/${EXTEN:1}@sip.discounttelecom.ru)
    
 
[LocalOnly]
    include => NSoftSipNet
    exten => _XXXX,1,Dial(SIP/${EXTEN:0}) 

 
[LocalAndPSTN]
    include => LocalOnly
    exten => _97XXXXXXX,1,Dial(SIP/8383${EXTEN:1}@sip.discounttelecom.ru)

 
[LocalAndPSTNAndMobile]
    include => LocalAndPSTN
    exten => _97XXXXXXXXX,1,Dial(SIP/{$EXTEN:1}@sip.discounttelecom.ru)
 
[LocalAndPSTNAndMobileAndLD]
    include => LocalAndPSTNAndMobile
    exten => _97XXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@sip.discounttelecom.ru)
    
;================== End CONTEXT ============================



 

;=================== Start VOICE MAIL ======================
    exten => 8000,1,VoiceMailMain
    exten => 8000,2,Hangup
;=================== End VOICE MAIL ========================
;=================== Start ALL USERS =======================
;    exten => 1000,1,Dial(SIP/1000,300,rtwW)			 
;    exten => 1000,2,Hangup					 
    
 
    exten => 1000,1,Dial(SIP/1000,8,rtwW)			 
    exten => 1000,2,Dial(SIP/79015582351@sip.discounttelecom.ru,300,rtwW)	 
    exten => 1000,3,Hangup					 

 
    exten => 1001,1,Dial(SIP/1001,8,rtwW)			 
    exten => 1001,2,Voicemail					 
    exten => 1001,3,Hangup					 

 
;    exten => 1000,1,GotoIfTime(9:00-18:00|mon-fri|*|*?4)	 
;    exten => 1000,2,Dial(SIP/1000,15,rtwW)			 
;    exten => 1000,3,Goto(8)					 
;    exten => 1000,4,Dial(SIP/1000.15,rtwW)			 
;    exten => 1000,5,Set(CALLERID(num)=3578378)			 
;    exten => 1000,6,Set(CALLERID(name)=my_astrerisk)		 
;    exten => 1000,7,Dial(SIP/79015582351@sip.discounttelecom.ru,300,Wj)	 
;    exten => 1000,8,Hangup					  
;==================== End ALL USERS ========================
входящая связь проходит. в сипнете с этим же конфигом, связь работает в обе стороны. сижу за натом порты проброшены реальник есть.
Вернуться к началу
ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение ded »

Не надо использовать синтаксис Dial(SIP/${EXTEN:1}@sip.discounttelecom.ru)

1) ping sip.discounttelecom.ru - может пинга нет!

2) Сделайте SIP peer
[discounttelecom]
type=peer
host=sip.discounttelecom.ru
username=
secret=

Он должен быть виден у вас по команде sip show peers

3) и звоните через него Dial(SIP/discounttelecom/${EXTEN:1})
Вернуться к началу
terra2039
Сообщения: 7
Зарегистрирован: 24 апр 2011, 12:36

Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение terra2039 »

ошибка осталась. вот дебаг.

Код: Выделить всё

Reliably Transmitting (NAT) to 192.168.0.101:60680:
OPTIONS sip:1000@192.168.0.101:60680;rinstance=67c641341740c265 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK6bca05cb;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.103>;tag=as165c3ec0
To: <sip:1000@192.168.0.101:60680;rinstance=67c641341740c265>
Contact: <sip:asterisk@192.168.0.103>
Call-ID: 282803db30e914db78c711325b016a69@192.168.0.103
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze2
Date: Wed, 25 May 2011 08:01:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.101:60680 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK6bca05cb;rport=5060
Contact: <sip:192.168.0.101:60680>
To: <sip:1000@192.168.0.101:60680;rinstance=67c641341740c265>;tag=ac7f654a
From: "asterisk"<sip:asterisk@192.168.0.103>;tag=as165c3ec0
Call-ID: 282803db30e914db78c711325b016a69@192.168.0.103
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '282803db30e914db78c711325b016a69@192.168.0.103' Method: OPTIONS

<--- SIP read from UDP:192.168.0.101:60680 --->



<------------->
[May 25 12:01:26] NOTICE[3094]: chan_sip.c:11655 sip_reregister:    -- Re-registration for  74959845383@sip.discounttelecom.ru
[May 25 12:01:27] NOTICE[3094]: chan_sip.c:18397 handle_response_register: Outbound Registration: Expiry for sip.discounttelecom.ru is 120 sec (Scheduling reregistration in 105 s)

<--- SIP read from UDP:192.168.0.101:60680 --->



<------------->
gw*CLI>
Disconnected from Asterisk server
root@gw:~# asterisk -r
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on gw (pid = 3082)
gw*CLI> sip set debug peer 1000
SIP Debugging Enabled for IP: 192.168.0.101:60680

<--- SIP read from UDP:192.168.0.101:60680 --->



<------------->

<--- SIP read from UDP:192.168.0.101:60680 --->
INVITE sip:979015582351@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:60680;branch=z9hG4bK-d8754z-ba3dd6437b47ec2a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1000@192.168.0.101:60680>
To: <sip:979015582351@192.168.0.103>
From: <sip:1000@192.168.0.103>;tag=730852f3
Call-ID: YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 410

v=0
o=- 12950787732535156 1 IN IP4 192.168.0.101
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.0.101
t=0 0
a=ice-ufrag:585e2d
a=ice-pwd:0e3d600e4d118fb06c0ecd52dae21f12
m=audio 50126 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.0.101 50126 typ host
a=candidate:1 2 UDP 659134 192.168.0.101 50127 typ host

<------------->
--- (13 headers 14 lines) ---
Sending to 192.168.0.101 : 60680 (no NAT)
Using INVITE request as basis request - YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
Found peer '1000' for '1000' from 192.168.0.101:60680

<--- Reliably Transmitting (NAT) to 192.168.0.101:60680 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.101:60680;branch=z9hG4bK-d8754z-ba3dd6437b47ec2a-1---d8754z-;received=192.168.0.101;rport=60680
From: <sip:1000@192.168.0.103>;tag=730852f3
To: <sip:979015582351@192.168.0.103>;tag=as2d354702
Call-ID: YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.9-2+squeeze2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="McConfig VoIP Server", nonce="466f20ef"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.101:60680 --->
ACK sip:979015582351@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:60680;branch=z9hG4bK-d8754z-ba3dd6437b47ec2a-1---d8754z-;rport
Max-Forwards: 70
To: <sip:979015582351@192.168.0.103>;tag=as2d354702
From: <sip:1000@192.168.0.103>;tag=730852f3
Call-ID: YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
CSeq: 1 ACK
Content-Length: 0


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

<--- SIP read from UDP:192.168.0.101:60680 --->
INVITE sip:979015582351@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:60680;branch=z9hG4bK-d8754z-169f9beb1e0b560f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1000@192.168.0.101:60680>
To: <sip:979015582351@192.168.0.103>
From: <sip:1000@192.168.0.103>;tag=730852f3
Call-ID: YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="1000",realm="McConfig VoIP Server",nonce="466f20ef",uri="sip:979015582351@192.168.0.103",response="bad1b5475b9b7438d7595442a29af3fe",algorithm=MD5
Content-Length: 410

v=0
o=- 12950787732535156 1 IN IP4 192.168.0.101
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.0.101
t=0 0
a=ice-ufrag:585e2d
a=ice-pwd:0e3d600e4d118fb06c0ecd52dae21f12
m=audio 50126 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.0.101 50126 typ host
a=candidate:1 2 UDP 659134 192.168.0.101 50127 typ host

<------------->
--- (14 headers 14 lines) ---
Sending to 192.168.0.101 : 60680 (NAT)
Using INVITE request as basis request - YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
Found peer '1000' for '1000' from 192.168.0.101:60680
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.101:50126
Looking for 979015582351 in LocalAndPSTNAndMobileAndLD (domain 192.168.0.103)
list_route: hop: <sip:1000@192.168.0.101:60680>

<--- Transmitting (NAT) to 192.168.0.101:60680 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.101:60680;branch=z9hG4bK-d8754z-169f9beb1e0b560f-1---d8754z-;received=192.168.0.101;rport=60680
From: <sip:1000@192.168.0.103>;tag=730852f3
To: <sip:979015582351@192.168.0.103>
Call-ID: YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9-2+squeeze2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:979015582351@192.168.0.103>
Content-Length: 0


<------------>

<--- Reliably Transmitting (NAT) to 192.168.0.101:60680 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.101:60680;branch=z9hG4bK-d8754z-169f9beb1e0b560f-1---d8754z-;received=192.168.0.101;rport=60680
From: <sip:1000@192.168.0.103>;tag=730852f3
To: <sip:979015582351@192.168.0.103>;tag=as050577c7
Call-ID: YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9-2+squeeze2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Network out of order
X-Asterisk-HangupCauseCode: 38


<------------>

<--- SIP read from UDP:192.168.0.101:60680 --->
ACK sip:979015582351@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:60680;branch=z9hG4bK-d8754z-169f9beb1e0b560f-1---d8754z-;rport
Max-Forwards: 70
To: <sip:979015582351@192.168.0.103>;tag=as050577c7
From: <sip:1000@192.168.0.103>;tag=730852f3
Call-ID: YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'YzgyYTZjNTM5ZjE5ZGQ3MjhjZDc3YjI2MWYzNTIxZDM.' Method: ACK
sip.conf

Код: Выделить всё

 [general]

    register = 74959845383:sip.discounttelecom.ru		 
    bindport = 5060					  
    bindaddr = 0.0.0.0					  
    context=default					 
    allowguest=yes 					 
    allowoverlap=no					 
    realm=McConfig VoIP Server				 
	
     						 
    srvlookup=yes					 
    language=ru 					 
    udpbindaddr=0.0.0.0
    tcpenable=no
    tcpbindaddr=0.0.0.0

 
    [authentication]
 
    [discounttelecom] 						 
    type=peer 					 
    insecure=invite					 
    secret=					 
    username=74959845383					 
    fromuser=74959845383 					 
    fromdomain=sip.discounttelecom.ru 				 
    host=sip.discounttelecom.ru 					 
    port=5060 						 
    dtmfmode=rfc2833 					 
    allow = all					 
    canreinvite=no 					 
						         

;============================ Users ===========================
[1000] ;ITCrowd						 
    qualify=yes						 
							 
    context=LocalAndPSTNAndMobileAndLD			 
    type=friend						 
    username=1000					 
    secret=						 
    nat=yes						 
    allow=all						 
    host=dynamic					 
    callerid="IT Crowd"					 
    callgroup=1						  
    pickupgroup=1					 
    language=ru						 
    canreinvite=yes					 
						         
    mailbox=1000@fantasyworld.org,1000			 

[1001] ;Support
    qualify=yes
    context=LocalAndPSTNAndMobileAndLD
    type=friend
    username=1001
    secret=1001
    nat=yes
    allow=all
    host=dynamic
    callerid="Support"
    callgroup=1
    pickupgroup=1
    language=ru
    canreinvite=yes
    mailbox=1001@fantasyworld.org,1001
extensions.conf

Код: Выделить всё

[general]
    priorityjumping=yes				 
    static=yes 					 
    writeprotect=no				 
    autofallthrough=yes				 
    clearglobalvars=no				 

 
[globals]
    DYNAMIC_FEATURES=automon
     
[default]
    include => 74959845383	
    include => OSoftSipNet
    include => NSoftSipNet			 
    include => LocalOnly			 
    include => LocalAndPSTN
    include => LocalAndPSTNAndMobile
    include => LocalAndPSTNAndMobileAndLD

;==================== Start SIPNET IN =======================
 
[74959845383]
    exten => s,1,Answer				 
    exten => s,2,Set(TIMEOUT(digit)=5)		 
    exten => s,3,Set(TIMEOUT(response)=10)	 
    exten => s,4,BackGround(test/1)		 
    exten => s,5,WaitExten(5)			 
    exten => s,6,Dial(SIP/1000,20,otmw)		 
    exten => s,7,Voicemail			 
    exten => s,8,Hangup				 
    exten => i,1,Dial(SIP/1000,300,otmw)
    exten => i,2,Hangup
    exten => _97XXXX,1,Goto(LocalAndPSTNAndMobileAndLD,${EXTEN},1)
;================== end SIPNET IN ==========================
;================== Start CONTEXT ==========================
  
[OSoftSipNet]
    exten => _9XXXXXXX,1,Dial(SIP/discounttelecom/${EXTEN:1})

 
[NSoftSipNet]
    include => OSoftSipNet
    exten => _900XXXXXXXX,1,Dial(SIP/discounttelecom/${EXTEN:1})

    
 
[LocalOnly]
    include => NSoftSipNet
    exten => _XXXX,1,Dial(SIP/${EXTEN:0}) 

 
[LocalAndPSTN]
    include => LocalOnly
    exten => _97XXXXXXX,1,Dial(SIP/discounttelecom/${EXTEN:1})

 
[LocalAndPSTNAndMobile]
    include => LocalAndPSTN
    exten => _97XXXXXXXXX,1,Dial(SIP/discounttelecom/${EXTEN:1})
 
[LocalAndPSTNAndMobileAndLD]
    include => LocalAndPSTNAndMobile
    exten => _979XXXXXXXXXX,1,Dial(SIP/discounttelecom/${EXTEN:1})
    
;================== End CONTEXT ============================



 

;=================== Start VOICE MAIL ======================
    exten => 8000,1,VoiceMailMain
    exten => 8000,2,Hangup
;=================== End VOICE MAIL ========================
;=================== Start ALL USERS =======================
;    exten => 1000,1,Dial(SIP/1000,300,rtwW)			 
;    exten => 1000,2,Hangup					 
    
 
    exten => 1000,1,Dial(SIP/1000,8,rtwW)			 
    exten => 1000,2,Dial(SIP/79015582351@sip.discounttelecom.ru,300,rtwW)	 
    exten => 1000,3,Hangup					 

 
    exten => 1001,1,Dial(SIP/1001,8,rtwW)			 
    exten => 1001,2,Voicemail					 
    exten => 1001,3,Hangup					 

 
;    exten => 1000,1,GotoIfTime(9:00-18:00|mon-fri|*|*?4)	 
;    exten => 1000,2,Dial(SIP/1000,15,rtwW)			 
;    exten => 1000,3,Goto(8)					 
;    exten => 1000,4,Dial(SIP/1000.15,rtwW)			 
;    exten => 1000,5,Set(CALLERID(num)=3578378)			 
;    exten => 1000,6,Set(CALLERID(name)=my_astrerisk)		 
;    exten => 1000,7,Dial(SIP/79015582351@sip.discounttelecom.ru,300,Wj)	 
;    exten => 1000,8,Hangup					  
;==================== End ALL USERS ========================
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ded
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Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение ded »

ded писал(а):1) ping sip.discounttelecom.ru - может пинга нет!
Пингуется?
ded писал(а):2) Сделайте SIP peer
[discounttelecom]Он должен быть виден у вас по команде sip show peers
Пир виден такой по команде?
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terra2039
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Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение terra2039 »

Код: Выделить всё

gw*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1000/1000                  192.168.0.101    D   N      57658    OK (45 ms)
1001/1001                  (Unspecified)    D   N      5060     UNKNOWN
discounttelecom/749598453  81.94.129.162               5060     Unmonitored
3 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 0 offline]

Код: Выделить всё

PING sip.discounttelecom.ru (81.94.129.162) 56(84) bytes of data.
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=1 ttl=59 time=3.64 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=2 ttl=59 time=4.53 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=3 ttl=59 time=4.09 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=4 ttl=59 time=4.04 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=5 ttl=59 time=6.49 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=6 ttl=59 time=14.0 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=7 ttl=59 time=3.52 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=8 ttl=59 time=4.08 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=9 ttl=59 time=3.78 ms
64 bytes from 162.129.94.81.runext.com (81.94.129.162): icmp_req=10 ttl=59 time=4.93 ms
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ded
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Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение ded »

ded писал(а):3) и звоните через него Dial(SIP/discounttelecom/${EXTEN:1})
что пишет при этом CLI и sip debug?
(не надо много закидывать, надо точно: строка Dial(SIP/discounttelecom/${EXTEN:1})
SIP Invite вашему провайдеру и SIP ответ от него. SIP дебаг вашего X-lite на 192.168.0.103 не нужно совсам (как Вы намусорили выше).
Нужен SIP debug peer discounttelecom

Видно было, что он отвечал (через * уже Х-лайту) - SIP/2.0 503 Service Unavailable
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terra2039
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Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение terra2039 »

дебаг от дисконттелеком

Код: Выделить всё

Scheduling destruction of SIP dialog '48beff4b623aab3749953b1f04b43dd4@127.0.1.1' in 32000 ms (Method: REGISTER)
[May 25 16:19:02] NOTICE[5674]: chan_sip.c:18397 handle_response_register: Outbound Registration: Expiry for sip.discounttelecom.ru is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '48beff4b623aab3749953b1f04b43dd4@127.0.1.1' Method: REGISTER
[May 25 16:20:01] WARNING[5837]: chan_sip.c:5581 sip_call: No audio format found to offer. Cancelling call to 74954513260
Scheduling destruction of SIP dialog '4a7825e41ba337fc66b8dfd3680b0263@sip.discounttelecom.ru' in 32000 ms (Method: INVITE)
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ded
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Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение ded »

No audio format found to offer отвечает Вам sip.discounttelecom.ru
Какое слово перевести?
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terra2039
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Зарегистрирован: 24 апр 2011, 12:36

Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение terra2039 »

да и так все ясно нужен кодек g711a-law провайдер его для передачи голоса использует.
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ded
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Re: dial_exec_full: Unable to create channel of type 'SIP' (

Сообщение ded »

Ага, а его как назло кто-то стырил из станции :)
Памагите! Нету g711a-law! Ни у кого нету, чтобы на станцию залить?
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