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

**terra2039** » 24 май 2011, 21:33

при исходящем вызове получаю ошибку.

Код: выделить все: dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)

sip.conf

Код: выделить все: [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 все: [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.

**ded** » 25 май 2011, 11:15

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})

**terra2039** » 25 май 2011, 13:28

ошибка осталась. вот дебаг.

Код: выделить все: 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 все: [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 все: [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 ========================

**ded** » 25 май 2011, 14:13

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?

**terra2039** » 25 май 2011, 16:25

Код: выделить все: 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 все: 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

**ded** » 25 май 2011, 16:48

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

**terra2039** » 25 май 2011, 17:21

дебаг от дисконттелеком

Код: выделить все: 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)

**ded** » 25 май 2011, 17:25

No audio format found to offer отвечает Вам sip.discounttelecom.ru
Какое слово перевести?

**terra2039** » 25 май 2011, 17:29

да и так все ясно нужен кодек g711a-law провайдер его для передачи голоса использует.

**ded** » 25 май 2011, 17:35

Ага, а его как назло кто-то стырил из станции

Памагите! Нету g711a-law! Ни у кого нету, чтобы на станцию залить?

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

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

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

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

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

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

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

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

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

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

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

Кто сейчас на форуме