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звонки проходят, у меня тишина

0

Доброго времени суток. Изменили схему регистрации. Вообщем сейчас цепляемся к провайдеру по логину и паролю. Забегая вперед скажу что роутер микротик рб750. Через софтфон (линфон,зоипер) регистрация проходит, звонки идут - довольны.

При звонке из офиса и в офис. Звонок проходит. Меня слышат, у меня тишина в трубке.

Если у кого-нибудь есть какие-нибудь мысли поделитесь пожалуйста

register => 567xx26:pass@sip.prov.ru/567xx26

bindport=5060
nat=yes
bindaddr=0.0.0.0
alwaysauthreject=yes
deny=0.0.0.0/0.0.0.0
permit=192.168.10.0/24
context=office                 ; Default context for incoming calls
allowguest=no                  ; Allow or reject guest calls (default is yes)
disallow=all
allow=alaw
localnet=192.168.10.0/255.255.255.0
externip=46.xx.64.53
language=ru



;1123;567xx26
[567XX26]
type=peer
callerid="567XX26" <567XX26>
fromuser=567XX26
host=sip.prov.ru
fromdomain=sip.prov.ru
context=office
disallow=all
allow=alaw
insecure=port,invite
nat=yes
dtmfmode=rfc2833
qualify=yes

[700]
type=friend
host=dynamic
username=700
secret=xx
nat=yes
canreinvite=no
context=office
callerid="700" <700>
disallow=all
allow=alaw
allow=ulaw
qualify=yes
dtmfmode=rfc2833

лог с астера

    <--- SIP read from UDP:192.168.10.241:5060 --->
INVITE sip:8965xxx2166@192.168.10.30 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;rport;branch=z9hG4bK1965116428
From: <sip:700@192.168.10.30>;tag=2028055494
To: <sip:8965xxx2166@192.168.10.30>
Call-ID: 197399179
CSeq: 20 INVITE
Contact: <sip:700@192.168.10.241>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Subject: Phone call
Content-Length:   310

v=0
o=700 2461 2461 IN IP4 192.168.0.15
s=Talk
c=IN IP4 192.168.0.15
t=0 0
m=audio 7078 RTP/AVP 8 112 111 110 3 0 101
a=rtpmap:112 speex/32000
a=fmtp:112 vbr=on
a=rtpmap:111 speex/16000
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
--- (13 headers 14 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 192.168.10.241 : 5060 (NAT)
Using INVITE request as basis request - 197399179
Found peer '700' for '700' from 192.168.10.241:5060
Found RTP audio format 8
Found RTP audio format 112
Found RTP audio format 111
Found RTP audio format 110
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format speex for ID 112
Found audio description format speex for ID 111
Found audio description format speex for ID 110
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0xa1e (gsm|ulaw|alaw|g726|speex|g726aal2)/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 192.168.0.15:7078
Looking for 8965xxx2166 in office (domain 192.168.10.30)
list_route: hop: <sip:700@192.168.10.241>

<--- Transmitting (NAT) to 192.168.10.241:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK1965116428;received=192.168.10.241;rport=5060
From: <sip:700@192.168.10.30>;tag=2028055494
To: <sip:8965xxx2166@192.168.10.30>
Call-ID: 197399179
CSeq: 20 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8965xxx2166@192.168.10.30>
Content-Length: 0


<------------>
    -- Executing [8965xxx2166@office:1] Dial("SIP/700-00000072", "SIP/567xx26/8965xxx2166") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 46.xx.64.53 port 18598
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 217.30.250.146:5060:
INVITE sip:8965xxx2166@217.30.250.146 SIP/2.0
Via: SIP/2.0/UDP 46.xx.64.53:5060;branch=z9hG4bK39594212;rport
Max-Forwards: 70
From: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
To: <sip:8965xxx2166@217.30.250.146>
Contact: <sip:567xx26@46.xx.64.53>
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Fri, 27 Sep 2013 08:22:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 894135055 894135055 IN IP4 46.xx.64.53
s=Asterisk PBX 1.6.2.9-2+squeeze3
c=IN IP4 46.xx.64.53
t=0 0
m=audio 18598 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called 567xx26/8965xxx2166

<--- SIP read from UDP:217.30.250.146:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 46.xx.64.53:5060;received=46.xx.64.53;branch=z9hG4bK39594212;rport
From: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
To: <sip:8965xxx2166@217.30.250.146>
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 102 INVITE


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

<--- SIP read from UDP:217.30.250.146:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 46.xx.64.53:5060;received=46.xx.64.53;branch=z9hG4bK39594212;rport
From: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
To: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,INFO,PRACK,UPDATE,OPTIONS,REGISTER,REFER,SUBSCRIBE,PUBLISH
Contact: <sip:8965xxx2166@217.30.250.146:5060;transport=udp>
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/567xx26-00000073 is ringing

<--- Transmitting (NAT) to 192.168.10.241:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK1965116428;received=192.168.10.241;rport=5060
From: <sip:700@192.168.10.30>;tag=2028055494
To: <sip:8965xxx2166@192.168.10.30>;tag=as053a5a93
Call-ID: 197399179
CSeq: 20 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8965xxx2166@192.168.10.30>
Content-Length: 0


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

<--- SIP read from UDP:217.30.250.146:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 46.xx.64.53:5060;received=46.xx.64.53;branch=z9hG4bK39594212;rport
From: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
To: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,INFO,PRACK,UPDATE,OPTIONS,REGISTER,REFER,SUBSCRIBE,PUBLISH
Contact: <sip:8965xxx2166@217.30.250.146:5060;transport=udp>
Content-Length: 221
Content-Type: application/sdp
Content-Disposition: session;handling=required

v=0
o=- 5084263 7756505 IN IP4 217.30.250.146
s=-
c=IN IP4 217.30.250.146
b=AS:64
t=0 0
m=audio 20156 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 217.30.250.146:20156
    -- SIP/567xx26-00000073 is making progress passing it to SIP/700-00000072
Audio is at 192.168.10.30 port 11266
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 192.168.10.241:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK1965116428;received=192.168.10.241;rport=5060
From: <sip:700@192.168.10.30>;tag=2028055494
To: <sip:8965xxx2166@192.168.10.30>;tag=as053a5a93
Call-ID: 197399179
CSeq: 20 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8965xxx2166@192.168.10.30>
Content-Type: application/sdp
Content-Length: 296

v=0
o=root 1727231804 1727231804 IN IP4 192.168.10.30
s=Asterisk PBX 1.6.2.9-2+squeeze3
c=IN IP4 192.168.10.30
t=0 0
m=audio 11266 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=ptime:20
a=sendrecv

<------------>
<------------>
Scheduling destruction of SIP dialog '8_f50d6d698e5952463c40_R@192.168.10.215' in 32000 ms (Method: REGISTER)
<------------>
Scheduling destruction of SIP dialog '8_f50d6d698e5952463c40_R@192.168.10.215' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:217.30.250.146:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 46.21.64.53:5060;received=46.21.64.53;branch=z9hG4bK39594212;rport
From: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
To: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,INFO,PRACK,UPDATE,OPTIONS,REGISTER,REFER,SUBSCRIBE,PUBLISH
Contact: <sip:8965xxx2166@217.30.250.146:5060;transport=udp>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/567xx26-00000073 is ringing
Reliably Transmitting (no NAT) to 127.0.0.1:4570:
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 46.21.64.53:5060;branch=z9hG4bK7360157c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@46.21.64.53>;tag=as296cd8df
To: <sip:127.0.0.1>
Contact: <sip:asterisk@46.21.64.53>
Call-ID: 7da53bf23bf3de4b447965ce74f5eb5f@46.21.64.53
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Fri, 27 Sep 2013 08:22:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<--- SIP read from UDP:217.30.250.146:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 46.21.64.53:5060;received=46.21.64.53;branch=z9hG4bK39594212;rport
From: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
To: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 102 INVITE
Accept: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,OPTIONS,REGISTER,REFER,SUBSCRIBE,MESSAGE,PUBLISH
Contact: "8965xxx2166" <sip:8965xxx2166@217.30.250.146:5060;transport=udp>
Supported: 100rel
Content-Length: 221
Content-Type: application/sdp
Content-Disposition: session;handling=required

v=0
o=- 5084263 7756505 IN IP4 217.30.250.146
s=-
c=IN IP4 217.30.250.146
b=AS:64
t=0 0
m=audio 20156 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (13 headers 12 lines) ---
list_route: hop: <sip:8965xxx2166@217.30.250.146:5060;transport=udp>
set_destination: Parsing <sip:8965xxx2166@217.30.250.146:5060;transport=udp> for address/port to send to
set_destination: set destination to 217.30.250.146, port 5060
Transmitting (NAT) to 217.30.250.146:5060:
ACK sip:8965xxx2166@217.30.250.146:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 46.21.64.53:5060;branch=z9hG4bK2b8b6c6a;rport
Max-Forwards: 70
From: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
To: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
Contact: <sip:567xx26@46.21.64.53>
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0
---
    -- SIP/567xx26-00000073 answered SIP/700-00000072
Audio is at 192.168.10.30 port 11266
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.10.241:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.15:5060;branch=z9hG4bK1965116428;received=192.168.10.241;rport=5060
From: <sip:700@192.168.10.30>;tag=2028055494
To: <sip:8965xxx2166@192.168.10.30>;tag=as053a5a93
Call-ID: 197399179
CSeq: 20 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:8965xxx2166@192.168.10.30>
Content-Type: application/sdp
Content-Length: 296

v=0
o=root 1727231804 1727231805 IN IP4 192.168.10.30
s=Asterisk PBX 1.6.2.9-2+squeeze3
c=IN IP4 192.168.10.30
t=0 0
m=audio 11266 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=ptime:20
a=sendrecv

<------------>
    -- Packet2Packet bridging SIP/700-00000072 and SIP/567xx26-00000073

<--- SIP read from UDP:192.168.10.241:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.10.241:5060 --->
ACK sip:8965xxx2166@192.168.10.30 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.15:5060;rport;branch=z9hG4bK987598064
From: <sip:700@192.168.10.30>;tag=2028055494
To: <sip:8965xxx2166@192.168.10.30>;tag=as053a5a93
Call-ID: 197399179
CSeq: 20 ACK
Contact: <sip:700@192.168.10.241>
Max-Forwards: 70
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0

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

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.10.215 : 1026 (NAT)


<------------>
Scheduling destruction of SIP dialog '8_f50d6d698e5952463c40_R@192.168.10.215' in 32000 ms (Method: REGISTER)
Retransmitting #1 (no NAT) to 127.0.0.1:4570:
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 46.21.64.53:5060;branch=z9hG4bK7360157c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@46.21.64.53>;tag=as296cd8df
To: <sip:127.0.0.1>
Contact: <sip:asterisk@46.21.64.53>
Call-ID: 7da53bf23bf3de4b447965ce74f5eb5f@46.21.64.53
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Fri, 27 Sep 2013 08:22:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Retransmitting #2 (no NAT) to 127.0.0.1:4570:
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 46.21.64.53:5060;branch=z9hG4bK7360157c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@46.21.64.53>;tag=as296cd8df
To: <sip:127.0.0.1>
Contact: <sip:asterisk@46.21.64.53>
Call-ID: 7da53bf23bf3de4b447965ce74f5eb5f@46.21.64.53
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Fri, 27 Sep 2013 08:22:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:217.30.250.146:5060 --->
BYE sip:567xx26@46.21.64.53 SIP/2.0
Via: SIP/2.0/UDP 217.30.250.146:5060;branch=z9hG4bKset9s300e0fgoislh5p0sd0008053.1
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 1226 BYE
From: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
To: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
Reason: Q.850;cause=16;text="Normal call clearing"
Max-Forwards: 69
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 217.30.250.146 : 5060 (NAT)

<--- Transmitting (NAT) to 217.30.250.146:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.30.250.146:5060;branch=z9hG4bKset9s300e0fgoislh5p0sd0008053.1;received=217.30.250.146
From: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
To: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 1226 BYE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
  == Spawn extension (office, 8965xxx2166, 1) exited non-zero on 'SIP/700-00000072'
Scheduling destruction of SIP dialog '197399179' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:700@192.168.10.241> for address/port to send to
set_destination: set destination to 192.168.10.241, port 5060
Reliably Transmitting (NAT) to 192.168.10.241:5060:
BYE sip:700@192.168.10.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK20bab024;rport
Max-Forwards: 70
From: <sip:8965xxx2166@192.168.10.30>;tag=as053a5a93
To: <sip:700@192.168.10.30>;tag=2028055494
Call-ID: 197399179
CSeq: 102 BYE
User-Agent: Asterisk
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

<--- SIP read from UDP:192.168.10.241:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK20bab024;rport=5060
From: <sip:8965xxx2166@192.168.10.30>;tag=as053a5a93
To: <sip:700@192.168.10.30>;tag=2028055494
Call-ID: 197399179
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 '0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146' Method: BYE
Really destroying SIP dialog '197399179' Method: ACK

<--- SIP read from UDP:217.30.250.146:5060 --->
BYE sip:567xx26@46.21.64.53 SIP/2.0
Via: SIP/2.0/UDP 217.30.250.146:5060;branch=z9hG4bKset9s300e0fgoislh5p0sd0008053.1
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 1226 BYE
From: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
To: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
Reason: Q.850;cause=16;text="Normal call clearing"
Max-Forwards: 69
Content-Length: 0

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

<--- Transmitting (NAT) to 217.30.250.146:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.30.250.146:5060;branch=z9hG4bKset9s300e0fgoislh5p0sd0008053.1;received=217.30.250.146
From: <sip:8965xxx2166@217.30.250.146>;tag=pdoylm66hd
To: "700" <sip:567xx26@217.30.250.146>;tag=as3c04ddf1
Call-ID: 0d94dffa03ff3ec91b99cb7876d2c91f@217.30.250.146
CSeq: 1226 BYE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
удалить закрыть спам изменить тег редактировать

спросил 2013-09-27 12:26:28 +0400

hoaxer Gravatar hoaxer
121 12 3 11

Comments

1

Мысли простые . Настройте корректно НАТ и всё будет работать.

zzuz ( 2013-09-27 14:28:00 +0400 )редактировать

ага спасибо кое-что убрал заработало. как допилю выложу правильный вариант. Еще раз спасибо zzuz

hoaxer ( 2013-09-27 15:24:49 +0400 )редактировать

1 Ответ

0

К слову, замечал за Линфоном, что он не хотел корректно работать даже внутри сети без NAT'a (когда хардфоны работали без проблем). Все заработало как надо, когда в Линфоне я таки указал, будто бы Линфон за NAT'ом и указал в виде внешнего IP адреса собственный адрес (адрес компьютера, на котором был Линфон).

ссылка удалить спам редактировать

ответил 2013-10-02 21:14:07 +0400

Lexus45 Gravatar Lexus45
270 3 3 3

Comments

Нормально Линфон работает , не нужно сочинять.

zzuz ( 2013-10-02 22:26:34 +0400 )редактировать

zzuz, значит с Twinkle перепутал.

Lexus45 ( 2013-10-03 14:39:16 +0400 )редактировать

Твинкл тоже нормально пашет, по крайней мере на линухе ))) Иногда использую его для "удалённого управления" серверами через телефон

asdev ( 2013-10-10 17:51:52 +0400 )редактировать

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Задан: 2013-09-27 12:26:28 +0400

Просмотрен: 367 раз

Обновлен: Oct 02 '13

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