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История изменений [назад]

нажмите, чтобы скрыть/показать версии 1
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спросил 2011-07-29 17:01:01 +0400

v0ip Gravatar v0ip flag of Russian Federation

http://v0ip.tk/

Mera =>SIP => Asterisk

Добрый день. И вновь проблемы с SIP. С Mera звонки направлены на астер, при попытке совершить звонок получаю ошибку 34 - (34)No circuit/channel available

При этом в логах астера с дебагом сипа, льется следующее: 

Reliably Transmitting (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '33a5cb5971a047be6222628d5f239767@10.100.200.25:5060' Method: OPTIONS

sip.conf

[mera]
type=friend
context=mera
host=10.100.200.1
qualify=yes
disallow=all
allow=g729
allow=g723

Короче уже три дня бьюсь. помогите, кто сталкивался.

нажмите, чтобы скрыть/показать версии 2
теги изменены
редактировать

обновил 2011-07-29 22:37:05 +0400

meral Gravatar meral flag of Ukraine

http://pro-sip.net/

Mera =>SIP => Asterisk

Добрый день. И вновь проблемы с SIP. С Mera звонки направлены на астер, при попытке совершить звонок получаю ошибку 34 - (34)No circuit/channel available

При этом в логах астера с дебагом сипа, льется следующее: 

Reliably Transmitting (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '33a5cb5971a047be6222628d5f239767@10.100.200.25:5060' Method: OPTIONS

sip.conf

[mera]
type=friend
context=mera
host=10.100.200.1
qualify=yes
disallow=all
allow=g729
allow=g723

Короче уже три дня бьюсь. помогите, кто сталкивался.

нажмите, чтобы скрыть/показать версии 3
No.3 Revision
редактировать

обновил 2011-07-29 22:37:05 +0400

meral Gravatar meral flag of Ukraine

http://pro-sip.net/

Mera =>SIP => Asterisk

Добрый день. И вновь проблемы с SIP. С Mera звонки направлены на астер, при попытке совершить звонок получаю ошибку 34 - (34)No circuit/channel available

При этом в логах астера с дебагом сипа, льется следующее: 

Reliably Transmitting (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK19e33a4b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as1e6a9880
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '68cbd3ca6ac968eb7cf8d72e2c967cf5@10.100.200.25:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 10.100.200.1:5060:
OPTIONS sip:10.100.200.1 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.25:5060;branch=z9hG4bK601f1d3e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.100.200.25>;tag=as20d01804
To: <sip:10.100.200.1>
Contact: <sip:asterisk@10.100.200.25:5060>
Call-ID: 33a5cb5971a047be6222628d5f239767@10.100.200.25:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.5.0
Date: Fri, 29 Jul 2011 12:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '33a5cb5971a047be6222628d5f239767@10.100.200.25:5060' Method: OPTIONS

тут должен быть лог звонка. но реально только сип пингы были..удалил

sip.conf

[mera]
type=friend
context=mera
host=10.100.200.1
qualify=yes
disallow=all
allow=g729
allow=g723

Короче уже три дня бьюсь. помогите, кто сталкивался.

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.