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Asterisk GUI + исходящие zadarma

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Здравствуйте!

Проблема с совершением исходящих звонков через Zadarma, звонок проходит и сразу же обрывается - слышно занято в обоих концах. Входящие нормально работают. users.conf почти совпадает с http://wiki.zadarma.com/index.php/Asterisk

В логах проскакивала ошибка, но сейчас её уже нет:

WARNING[15041] chan_sip.c: Invalid contact uri  (missing sip: or sips:), attempting to use anyway
[Mar 18 16:25:56] WARNING[15041] chan_sip.c: Invalid URI: parse_uri failed to acquire hostport
[Mar 18 16:27:30] NOTICE[15041] chan_sip.c: Call from '1023' (192.168.1.51:5062) to extension '8495' rejected because extension not found in context 'DLPN_zadarDP'.
[Mar 18 16:28:10] WARNING[15041] chan_sip.c: Invalid contact uri  (missing sip: or sips:), attempting to use anyway
[Mar 18 16:28:10] WARNING[15041] chan_sip.c: Invalid URI: parse_uri failed to acquire hostport
Что может быть. В астериске новичок, все настраивалось через GUI-веб интерфейс. С другого провайдера сип работают как входящие так и исходящие.

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Тут проблема явно не в zadarma, из логов видно, что астериск пытается позвонить на номер 8495, который, очевидно, не существует.

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Для подробного ответа рекомендую обратится в службу поддержки, я так-же не специалист в asterisk.

 

P.S.: Насколько Хватает логики и английского: "Call from '1023'.. to extension '8495' rejected because extension not found in context 'DLPN_zadarDP'." Судя по этой строке звонок к Zadarma и не отправляли, оно оборвался в астериске т.к. либо не верно набрали номер либо не задали такой экстеншин в "context DLPN_zadarDP"

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Для подробного ответа рекомендую обратится в службу поддержки, я так-же не специалист в asterisk.

 

P.S.: Насколько Хватает логики и английского: "Call from '1023'.. to extension '8495' rejected because extension not found in context 'DLPN_zadarDP'." Судя по этой строке звонок к Zadarma и не отправляли, оно оборвался в астериске т.к. либо не верно набрали номер либо не задали такой экстеншин в "context DLPN_zadarDP"

В какую такую службу поддержки? Звонок, повторяю проходит, но срывается когда берут трубку - занято сразу. Да, это старые логи, вот полные:

<--- Transmitting (NAT) to 192.168.1.45:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK22829;received=192.168.1.45;rport=5060
From: <sip:100@192.168.10.30>;tag=18875
To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc
Call-ID: 4634
CSeq: 21 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74955554422@192.168.10.30:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 779405766 779405766 IN IP4 192.168.10.30
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.10.30
t=0 0
m=audio 12670 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:5.9.108.25:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7e96efdd;received=62.205.186.142;rport=28988
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as3086b515
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="sip.zadarma.com", nonce="14f867d0"
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:74955554422@176.9.145.115:5060> for address/port to send to
set_destination: set destination to 176.9.145.115:5060
Transmitting (no NAT) to 176.9.145.115:5060:
ACK sip:74955554422@176.9.145.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7d665df1
Max-Forwards: 70
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72
Contact: <sip:25638@192.168.10.30:5060>
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- SIP read from UDP:5.9.108.25:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7e96efdd;received=62.205.186.142;rport=28988
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as3086b515
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="sip.zadarma.com", nonce="14f867d0"
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:74955554422@176.9.145.115:5060> for address/port to send to
set_destination: set destination to 176.9.145.115:5060
Transmitting (no NAT) to 176.9.145.115:5060:
ACK sip:74955554422@176.9.145.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7d665df1
Max-Forwards: 70
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72
Contact: <sip:25638@192.168.10.30:5060>
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- SIP read from UDP:176.9.145.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7d665df1;received=62.205.186.142;rport=28988
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 104 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 324

v=0
o=root 684333766 684333767 IN IP4 176.9.145.115
s=Zadarma Voip
c=IN IP4 176.9.145.115
t=0 0
m=audio 11364 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99

<------------->
--- (11 headers 15 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
[Mar 18 22:47:54] WARNING[24299]: chan_sip.c:8949 process_sdp: ignoring 'video' media offer because port number is zeroCapabilities: us - 0x20080e (gsm|ulaw|alaw|g726|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 176.9.145.115:11364
Peer doesn't provide video
list_route: no route
[Mar 18 22:47:54] WARNING[24299]: chan_sip.c:13880 __set_address_from_contact: Invalid contact uri  (missing sip: or sips:), attempting to use anyway
[Mar 18 22:47:54] WARNING[24299]: chan_sip.c:13893 __set_address_from_contact: Invalid URI: parse_uri failed to acquire hostport
Transmitting (no NAT) to 176.9.145.115:5060:
ACK sip:74955554422@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK6829626b
Max-Forwards: 70
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72
Contact: <sip:25638@192.168.10.30:5060>
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Reliably Transmitting (no NAT) to 176.9.145.115:5060:
BYE sip:74955554422@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK46a996c9
Max-Forwards: 70
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 105 BYE
User-Agent: Asterisk PBX
Authorization: Digest username="25638", realm="sip.zadarma.com", algorithm=MD5, uri="sip:74955554422@sip.zadarma.com", nonce="562be498", response="d744ae4624301e41230c42ebf58ab852"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
Scheduling destruction of SIP dialog '6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com' in 32000 ms (Method: INVITE)
	-- SIP/Zadarma-00000005 answered SIP/100-00000004
Audio is at 12670
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.1.45:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK22829;received=192.168.1.45;rport=5060
From: <sip:100@192.168.10.30>;tag=18875
To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc
Call-ID: 4634
CSeq: 21 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74955554422@192.168.10.30:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 779405766 779405767 IN IP4 192.168.10.30
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.10.30
t=0 0
m=audio 12670 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:176.9.145.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK46a996c9;received=62.205.186.142;rport=28988
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 105 BYE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 176.9.145.115:5060:
BYE sip:74955554422@sip.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK4de0c041
Max-Forwards: 70
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 106 BYE
User-Agent: Asterisk PBX
Authorization: Digest username="25638", realm="sip.zadarma.com", algorithm=MD5, uri="sip:74955554422@sip.zadarma.com", nonce="562be498", response="d744ae4624301e41230c42ebf58ab852"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/100-00000004' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_DialPlan1, 74955554422, 1) exited non-zero on 'SIP/100-00000004'
Scheduling destruction of SIP dialog '4634' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:176.9.145.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK4de0c041;received=62.205.186.142;rport=28988
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as6d776e72
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 106 BYE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com' Method: INVITE
Retransmitting #1 (NAT) to 192.168.1.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK22829;received=192.168.1.45;rport=5060
From: <sip:100@192.168.10.30>;tag=18875
To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc
Call-ID: 4634
CSeq: 21 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74955554422@192.168.10.30:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 779405766 779405767 IN IP4 192.168.10.30
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.10.30
t=0 0
m=audio 12670 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.43:5060 --->
jaK
<------------->
Retransmitting #2 (NAT) to 192.168.1.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK22829;received=192.168.1.45;rport=5060
From: <sip:100@192.168.10.30>;tag=18875
To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc
Call-ID: 4634
CSeq: 21 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74955554422@192.168.10.30:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 779405766 779405767 IN IP4 192.168.10.30
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.10.30
t=0 0
m=audio 12670 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.45:5060 --->
ACK sip:74955554422@192.168.10.30:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.45:5060;rport;branch=z9hG4bK507
From: <sip:100@192.168.10.30>;tag=18875
To: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc
Call-ID: 4634
CSeq: 21 ACK
Contact: <sip:bulat@192.168.1.45>
Max-Forwards: 70
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:bulat@19119> for address/port to send to
set_destination: set destination to 0.0.74.175:5060
Reliably Transmitting (NAT) to 192.168.1.45:5060:
BYE sip:bulat@19119 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK4c5c2f2d;rport
Max-Forwards: 70
From: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc
To: <sip:100@192.168.10.30>;tag=18875
Call-ID: 4634
CSeq: 102 BYE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="100", realm="asterisk", algorithm=MD5, uri="sip:192.168.10.30", nonce="", response="8d3278878ce074848ab411f87e7c7c60"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog '4634' in 32000 ms (Method: ACK)

<--- SIP read from UDP:192.168.1.45:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK4c5c2f2d;rport=5060
From: <sip:74955554422@192.168.10.30>;tag=as1bf20bfc
To: <sip:100@192.168.10.30>;tag=18875
Call-ID: 4634
CSeq: 102 BYE
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '4634' Method: ACK

<--- SIP read from UDP:192.168.1.45:5060 --->
jaK
<------------->

<--- SIP read from UDP:5.9.108.25:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK7e96efdd;received=62.205.186.142;rport=28988
From: "asterisk" <sip:25638@sip.zadarma.com>;tag=as001ce4ad
To: <sip:74955554422@sip.zadarma.com>;tag=as3086b515
Call-ID: 6edfbaa922cf692c7d0a14157e1fd347@sip.zadarma.com
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="sip.zadarma.com", nonce="14f867d0"
Content-Length: 0


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

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