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t.38 Fax & * 1.4.18.1

Native bridging и... НИЧЕГО!
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contramundum
Сообщений: 89

Re: t.38 Fax & * 1.4.18.1

Firewall не поднят (во всяком случае службы ip6tables и iptables не подняты). Звоники идут без проблем, и с самого факса тоже только что звонили в город - всё ok.

[2008-10-30 14:42:56] DEBUG[3176] chan_sip.c: Peer doesn't provide T.38 UDPTL
[2008-10-30 14:42:56] DEBUG[3176] chan_sip.c: Strict routing enforced for session 6b72597155df44fd18c9440e2750254b@192.168.100.10
[2008-10-30 14:42:56] DEBUG[4121] chan_sip.c: Strict routing enforced for session 6b72597155df44fd18c9440e2750254b@192.168.100.10
[2008-10-30 14:42:56] DEBUG[3176] chan_sip.c: Strict routing enforced for session 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
[2008-10-30 14:42:56] DEBUG[3176] chan_sip.c: Peer doesn't provide T.38 UDPTL
[2008-10-30 14:42:56] DEBUG[3176] chan_sip.c: Strict routing enforced for session 6b72597155df44fd18c9440e2750254b@192.168.100.10
[2008-10-30 14:42:56] DEBUG[3176] chan_sip.c: Peer doesn't provide T.38 UDPTL
[2008-10-30 14:42:56] DEBUG[3176] chan_sip.c: Strict routing enforced for session 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
[2008-10-30 14:43:24] DEBUG[3176] chan_sip.c: Peer T.38 UDPTL is at port 192.168.100.20:16384
[2008-10-30 14:43:24] DEBUG[3176] chan_sip.c: Our T38 capability = (3856), peer T38 capability (16160), joint T38 capability (3872)
[2008-10-30 14:43:24] DEBUG[3176] chan_sip.c: Strict routing enforced for session 6b72597155df44fd18c9440e2750254b@192.168.100.10
[2008-10-30 14:43:24] DEBUG[3176] chan_sip.c: T.38 UDPTL is at 192.168.100.10 port 4599
[2008-10-30 14:43:24] DEBUG[3176] chan_sip.c: Our T38 capability (3856), peer T38 capability (3872), joint capability (3872)
[2008-10-30 14:43:25] DEBUG[3176] chan_sip.c: Peer T.38 UDPTL is at port 192.168.100.21:9008
[2008-10-30 14:43:25] DEBUG[3176] chan_sip.c: Our T38 capability = (3856), peer T38 capability (3872), joint T38 capability (3872)
[2008-10-30 14:43:25] DEBUG[3176] chan_sip.c: T.38 UDPTL is at 192.168.100.10 port 4192
[2008-10-30 14:43:25] DEBUG[3176] chan_sip.c: Our T38 capability (3856), peer T38 capability (3872), joint capability (3872)
[2008-10-30 14:43:25] DEBUG[3176] chan_sip.c: Strict routing enforced for session 6b72597155df44fd18c9440e2750254b@192.168.100.10
[2008-10-30 14:43:56] DEBUG[4121] chan_sip.c: Strict routing enforced for session 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
[2008-10-30 14:43:57] DEBUG[3176] chan_sip.c: Peer doesn't provide T.38 UDPTL
[2008-10-30 14:43:57] DEBUG[3176] chan_sip.c: Strict routing enforced for session 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
[2008-10-30 14:43:57] DEBUG[3176] chan_sip.c: Strict routing enforced for session 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20

100.10 - Asterisk
100.20 - Micronet с аналоговым факсом на одном из портов



<------------->
[2008-10-30 14:43:01] VERBOSE[3176] logger.c: --- (12 headers 0 lines) ---
[2008-10-30 14:43:01] VERBOSE[3176] logger.c: Using latest REGISTER request as basis request
[2008-10-30 14:43:01] VERBOSE[3176] logger.c: Sending to 192.168.100.21 : 5060 (no NAT)
[2008-10-30 14:43:01] VERBOSE[3176] logger.c:
<--- Transmitting (no NAT) to 192.168.100.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK8723bfbd7ab28040;received=192.168.100.21
From: <sip:0000@192.168.100.10>;tag=3de922ec-690150
To: <sip:0000@192.168.100.10>
Call-ID: D1B9-7526-466901503DEEEC84301C-080@SipHost
CSeq: 20 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0000@192.168.100.10>
Content-Length: 0


<------------>
[2008-10-30 14:43:01] VERBOSE[3176] logger.c:
<--- Transmitting (no NAT) to 192.168.100.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK8723bfbd7ab28040;received=192.168.100.21
From: <sip:0000@192.168.100.10>;tag=3de922ec-690150
To: <sip:0000@192.168.100.10>;tag=as28e119a7
Call-ID: D1B9-7526-466901503DEEEC84301C-080@SipHost
CSeq: 20 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 600
Contact: <sip:0000@192.168.100.21:5060>;expires=600
Date: Thu, 30 Oct 2008 11:43:01 GMT
Content-Length: 0


<------------>
[2008-10-30 14:43:01] VERBOSE[3176] logger.c: Scheduling destruction of SIP dialog 'D1B9-7526-466901503DEEEC84301C-080@SipHost' in 32000 ms (Method: REGISTER)
[2008-10-30 14:43:24] VERBOSE[3176] logger.c:
<--- SIP read from 192.168.100.20:5060 --->
INVITE sip:9XXXXXXX@192.168.100.10 SIP/2.0
From: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
To: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 3 INVITE
Via: SIP/2.0/UDP 192.168.100.20:5060;branch=z9hG4bK-14ce-51667a-6363
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (2asipfxs.114)
Contact: <sip:33@192.168.100.20:5060>
Proxy-Authorization: Digest username="33", realm="asterisk", nonce="0cd8886f", uri="sip:9XXXXXXX@192.168.100.10", response="7a1cd0e14bce80772e20db193127e4d4", algorithm=MD5
Content-Type: application/sdp
Content-Length: 382

v=0
o=ata_sip_1 45721 45721 IN IP4 192.168.100.20
s=Audio Session
i=Audio Session
c=IN IP4 192.168.100.20
t=0 0
m=image 16384 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: --- (13 headers 16 lines) ---
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: Sending to 192.168.100.20 : 5060 (no NAT)
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: Got T.38 offer in SDP in dialog 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2008-10-30 14:43:24] VERBOSE[3176] logger.c:
<--- Transmitting (no NAT) to 192.168.100.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.20:5060;branch=z9hG4bK-14ce-51667a-6363;received=192.168.100.20
From: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
To: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9XXXXXXX@192.168.100.10>
Content-Length: 0


<------------>
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: set_destination: Parsing <sip:0000@192.168.100.21:5060> for address/port to send to
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: set_destination: set destination to 192.168.100.21, port 5060
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: Reliably Transmitting (no NAT) to 192.168.100.21:5060:
INVITE sip:0000@192.168.100.21:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK6ce4e86b;rport
From: "33-FAX" <sip:33@192.168.100.10>;tag=as2721b3ec
To: <sip:XXXXXXX@192.168.100.21>;tag=edc88b21-690145
Contact: <sip:33@192.168.100.10>
Call-ID: 6b72597155df44fd18c9440e2750254b@192.168.100.10
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 352

v=0
o=root 2830 2832 IN IP4 192.168.100.20
s=session
c=IN IP4 192.168.100.20
t=0 0
m=image 16384 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:316
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

---
[2008-10-30 14:43:24] VERBOSE[3176] logger.c:
<--- SIP read from 192.168.100.21:5060 --->
SIP/2.0 100 Trying
v:SIP/2.0/UDP 192.168.100.10:5060;rport;branch=z9hG4bK6ce4e86b
f:"33-FAX" <sip:33@192.168.100.10>;tag=as2721b3ec
t:<sip:XXXXXXX@192.168.100.21>;tag=edc88b21-690145
i:6b72597155df44fd18c9440e2750254b@192.168.100.10
CSeq:104 INVITE
c:application/sdp
l:0


<------------->
[2008-10-30 14:43:24] VERBOSE[3176] logger.c: --- (8 headers 0 lines) ---
[2008-10-30 14:43:25] VERBOSE[3176] logger.c:
<--- SIP read from 192.168.100.21:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
v:SIP/2.0/UDP 192.168.100.10:5060;rport;branch=z9hG4bK6ce4e86b
f:"33-FAX" <sip:33@192.168.100.10>;tag=as2721b3ec
t:<sip:XXXXXXX@192.168.100.21>;tag=edc88b21-690145
i:6b72597155df44fd18c9440e2750254b@192.168.100.10
CSeq:104 INVITE
m:<sip:0000@192.168.100.21:5060>
User-Agent:dlink 12-38-16928528-0.9.5.1.735-PNP163
c:application/sdp
l:367

v=0
o=0005 1797369660 1797369660 IN IP4 192.168.100.21
s=Session SDP
c=IN IP4 192.168.100.21
t=0 0
m=image 9008 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:316
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: --- (11 headers 15 lines) ---
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: Got T.38 offer in SDP in dialog 6b72597155df44fd18c9440e2750254b@192.168.100.10
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 6b72597155df44fd18c9440e2750254b@192.168.100.10
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2008-10-30 14:43:25] VERBOSE[3176] logger.c:
<--- Reliably Transmitting (no NAT) to 192.168.100.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.20:5060;branch=z9hG4bK-14ce-51667a-6363;received=192.168.100.20
From: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
To: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9XXXXXXX@192.168.100.10>
Content-Type: application/sdp
Content-Length: 351

v=0
o=root 2830 2832 IN IP4 192.168.100.21
s=session
c=IN IP4 192.168.100.21
t=0 0
m=image 9008 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:316
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

<------------>
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: set_destination: Parsing <sip:0000@192.168.100.21:5060> for address/port to send to
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: set_destination: set destination to 192.168.100.21, port 5060
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: Transmitting (no NAT) to 192.168.100.21:5060:
ACK sip:0000@192.168.100.21:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK1af10fbc;rport
From: "33-FAX" <sip:33@192.168.100.10>;tag=as2721b3ec
To: <sip:XXXXXXX@192.168.100.21>;tag=edc88b21-690145
Contact: <sip:33@192.168.100.10>
Call-ID: 6b72597155df44fd18c9440e2750254b@192.168.100.10
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[2008-10-30 14:43:25] VERBOSE[3176] logger.c:
<--- SIP read from 192.168.100.20:5060 --->
ACK sip:9XXXXXXX@192.168.100.10 SIP/2.0
From: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
To: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 3 ACK
Via: SIP/2.0/UDP 192.168.100.20:5060;branch=z9hG4bK-14ce-51680f-2b60
Max-Forwards: 70
User-Agent: FXS_GW (2asipfxs.114)
Contact: <sip:33@192.168.100.20:5060>
Proxy-Authorization: Digest username="33", realm="asterisk", nonce="0cd8886f", uri="sip:9XXXXXXX@192.168.100.10", response="7a1cd0e14bce80772e20db193127e4d4", algorithm=MD5
Content-Length: 0


<------------->
[2008-10-30 14:43:25] VERBOSE[3176] logger.c: --- (11 headers 0 lines) ---
[2008-10-30 14:43:28] VERBOSE[3176] logger.c: Really destroying SIP dialog 'D1B9-7526-46690145B82BE71A75D7-073@SipHost' Method: REGISTER
[2008-10-30 14:43:28] VERBOSE[3176] logger.c: Really destroying SIP dialog 'D1B9-7526-4669014664203E30689E-074@SipHost' Method: REGISTER
[2008-10-30 14:43:29] VERBOSE[3176] logger.c: Really destroying SIP dialog 'D1B9-7526-4669014793B4EF4E6D88-075@SipHost' Method: REGISTER
[2008-10-30 14:43:29] VERBOSE[3176] logger.c: Really destroying SIP dialog 'D1B9-7526-4669014785497CC29679-076@SipHost' Method: REGISTER
[2008-10-30 14:43:32] VERBOSE[3176] logger.c: Really destroying SIP dialog 'D1B9-7526-466901506ADB5EA5F774-077@SipHost' Method: REGISTER
[2008-10-30 14:43:32] VERBOSE[3176] logger.c: Really destroying SIP dialog 'D1B9-7526-466901504C279AA4BA8E-078@SipHost' Method: REGISTER
[2008-10-30 14:43:33] VERBOSE[3176] logger.c: Really destroying SIP dialog 'D1B9-7526-4669015076E3D12528FC-079@SipHost' Method: REGISTER
[2008-10-30 14:43:33] VERBOSE[3176] logger.c: Really destroying SIP dialog 'D1B9-7526-466901503DEEEC84301C-080@SipHost' Method: REGISTER
[2008-10-30 14:43:56] VERBOSE[3176] logger.c:
<--- SIP read from 192.168.100.21:5060 --->
BYE sip:33@192.168.100.10 SIP/2.0
v:SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bKe538ecee45e0668c
f:<sip:XXXXXXX@192.168.100.21>;tag=edc88b21-690145
t:"33-FAX" <sip:33@192.168.100.10>;tag=as2721b3ec
i:6b72597155df44fd18c9440e2750254b@192.168.100.10
CSeq:21 BYE
m:<sip:0000@192.168.100.21:5060>
Max-Forwards:70
User-Agent:dlink 12-38-16928528-0.9.5.1.735-PNP163
l:0


<------------->
[2008-10-30 14:43:56] VERBOSE[3176] logger.c: --- (10 headers 0 lines) ---
[2008-10-30 14:43:56] VERBOSE[3176] logger.c: Sending to 192.168.100.21 : 5060 (no NAT)
[2008-10-30 14:43:56] VERBOSE[3176] logger.c:
<--- Transmitting (no NAT) to 192.168.100.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bKe538ecee45e0668c;received=192.168.100.21
From: <sip:XXXXXXX@192.168.100.21>;tag=edc88b21-690145
To: "33-FAX" <sip:33@192.168.100.10>;tag=as2721b3ec
Call-ID: 6b72597155df44fd18c9440e2750254b@192.168.100.10
CSeq: 21 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:33@192.168.100.10>
Content-Length: 0


<------------>
[2008-10-30 14:43:56] VERBOSE[4121] logger.c: set_destination: Parsing <sip:33@192.168.100.20:5060> for address/port to send to
[2008-10-30 14:43:56] VERBOSE[4121] logger.c: set_destination: set destination to 192.168.100.20, port 5060
[2008-10-30 14:43:56] VERBOSE[4121] logger.c: Audio is at 192.168.100.10 port 11782
[2008-10-30 14:43:56] VERBOSE[4121] logger.c: Adding codec 0x4 (ulaw) to SDP
[2008-10-30 14:43:56] VERBOSE[4121] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[2008-10-30 14:43:56] VERBOSE[4121] logger.c: Reliably Transmitting (no NAT) to 192.168.100.20:5060:
INVITE sip:33@192.168.100.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK152ba9de;rport
From: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
To: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
Contact: <sip:9XXXXXXX@192.168.100.10>
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2830 2833 IN IP4 192.168.100.10
s=session
c=IN IP4 192.168.100.10
t=0 0
m=audio 11782 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[2008-10-30 14:43:56] VERBOSE[4121] logger.c: == Spawn extension (office, 9XXXXXXX, 12) exited non-zero on 'SIP/33-081e8ea0'
[2008-10-30 14:43:56] VERBOSE[4121] logger.c: Scheduling destruction of SIP dialog '570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20' in 32000 ms (Method: ACK)
[2008-10-30 14:43:57] VERBOSE[3176] logger.c:
<--- SIP read from 192.168.100.20:5060 --->
SIP/2.0 200 OK
From: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
To: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.100.10:5060;rport=5060;branch=z9hG4bK152ba9de
Supported: replaces
User-Agent: FXS_GW (2asipfxs.114)
Contact: <sip:33@192.168.100.20:5060>
Content-Type: application/sdp
Content-Length: 255

v=0
o=FXS_GW 12367 0 IN IP4 192.168.100.20
s=Audio Session
i=Audio Session
c=IN IP4 192.168.100.20
t=0 0
m=audio 16384 RTP/AVP 0 101
a=ptime:60
a=fmtp:101 0-11
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000

<------------->
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: --- (11 headers 12 lines) ---
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Found RTP audio format 0
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Found RTP audio format 101
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Peer audio RTP is at port 192.168.100.20:16384
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Found audio description format PCMU for ID 0
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Found audio description format telephone-event for ID 101
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Peer audio RTP is at port 192.168.100.20:16384
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: list_route: hop: <sip:33@192.168.100.20:5060>
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: set_destination: Parsing <sip:33@192.168.100.20:5060> for address/port to send to
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: set_destination: set destination to 192.168.100.20, port 5060
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Transmitting (no NAT) to 192.168.100.20:5060:
ACK sip:33@192.168.100.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK6f2525d3;rport
From: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
To: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
Contact: <sip:9XXXXXXX@192.168.100.10>
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: set_destination: Parsing <sip:33@192.168.100.20:5060> for address/port to send to
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: set_destination: set destination to 192.168.100.20, port 5060
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Reliably Transmitting (no NAT) to 192.168.100.20:5060:
BYE sip:33@192.168.100.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK40a13d3d;rport
From: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
To: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Scheduling destruction of SIP dialog '570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20' in 32000 ms (Method: ACK)
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Really destroying SIP dialog '6b72597155df44fd18c9440e2750254b@192.168.100.10' Method: BYE
[2008-10-30 14:43:57] VERBOSE[3176] logger.c:
<--- SIP read from 192.168.100.20:5060 --->
SIP/2.0 200 OK
From: <sip:9XXXXXXX@192.168.100.10;user=phone>;tag=as2fb0795c
To: "33"<sip:33@192.168.100.10>;tag=c0a86414-13c4-14a5-50c76f-3b48
Call-ID: 570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20
CSeq: 104 BYE
Via: SIP/2.0/UDP 192.168.100.10:5060;rport=5060;branch=z9hG4bK40a13d3d
Supported: replaces
User-Agent: FXS_GW (2asipfxs.114)
Content-Length: 0


<------------->
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: --- (9 headers 0 lines) ---
[2008-10-30 14:43:57] VERBOSE[3176] logger.c: Really destroying SIP dialog '570d9c-c0a86414-13c4-14a5-50c75b-1a61@192.168.100.20' Method: ACK

в логах вроде все нормально
2008-10-30 15:24

contramundum
Сообщений: 89

Re: t.38 Fax & * 1.4.18.1

за логи огромные уж извините, просто не знал, что конкретно в них искать...
33 - номер факса на *
9ХХХХХХХ и ХХХХХХХ - городской номер удаленного факса
100.10 - *
100.20 - Micronet
100.21 - D-Link
2008-10-30 15:28

litnimax
Avatara of litnimax
Откуда: Москва
Сообщений: 3421

Re: t.38 Fax & * 1.4.18.1

Попробуйте сделать insecure=invite.
http://pbxware.ru - все для Asterisk! || Switchvox - сделано на Asterisk! Подробности на http://switchvox.ru
2008-10-30 16:17

Evgen
Откуда: Саратов
Сообщений: 414

Re: t.38 Fax & * 1.4.18.1

Непонятно, по какой причине появляется вот это:

[2008-10-30 14:43:56] VERBOSE[3176] logger.c:
<--- SIP read from 192.168.100.21:5060 --->
BYE sip:33@192.168.100.10 SIP/2.0
v:SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bKe538ecee45e0668c
f:<sip:XXXXXXX@192.168.100.21>;tag=edc88b21-690145
t:"33-FAX" <sip:33@192.168.100.10>;tag=as2721b3ec
i:6b72597155df44fd18c9440e2750254b@192.168.100.10
CSeq:21 BYE
m:<sip:0000@192.168.100.21:5060>
Max-Forwards:70
User-Agent:dlink 12-38-16928528-0.9.5.1.735-PNP163
l:0

Запустите во время попытки передачи факса команду udptl debug
+7(925)140-7438
2008-10-30 17:44

dimas
Сообщений: 866

Re: t.38 Fax & * 1.4.18.1

А чисто из любопытства - вы пробовали micronet напрямую с dlink без астериска соединить? Ходят ли там факсы? А то если верить логу, ваши стороны прекрасно договариваются друг с другом на T38 UDPTL _в обход_ астериска. Так что что там дальше между ними происходит науке неизвестно - 30 сек они занимаются каким-то сексом напрямую а потом уже жалуются на несудьбу...
2008-10-30 17:55

contramundum
Сообщений: 89

Re: t.38 Fax & * 1.4.18.1

insecure=invite не совсем понял куда его пихать, поэтому вписал в sip.cof в [general] и в секцию факса... не помогло...

вот еще логи Микронета:
******** Line : 1, Start Inviting ********
strDes To:<sip:9ХХХХХХХ@192.168.100.10;user=phone>, strOri From:"33"<sip:33@192.
168.100.10>
1-RvSipCallLegMgrCreateCallLeg() ok!
1-RVSIP_CALL_LEG_STATE_INVITING
Sending RVSIP_METHOD_ACK1-RVSIP_CALL_LEG_STATE_UNAUTHENTICATED
1-RvSipCallLegAuthenticate
1-RVSIP_CALL_LEG_STATE_INVITING
Sending RVSIP_METHOD_ACK1-RVSIP_CALL_LEG_STATE_CONNECTED
1-RVSIP_CALL_LEG_MODIFY_STATE_IDLE
Selected codec g711Ulaw64k
1-RVSIP_CALL_LEG_MODIFY_STATE_REINVITE_RCVD
1-RVSIP_CALL_LEG_MODIFY_STATE_REINVITE_RESPONSE_SENT
AppCallLegModifyRequestRcvdEvHandler
Error!!! The Response Msg Received doesn't has Msg Body
1-RVSIP_CALL_LEG_MODIFY_STATE_IDLE
Selected codec g711Ulaw64k
CID-0 T.38 EV_DETECT_FAX ....
[vp_detect_fax] 1-RvSipCallLegModify
1-RVSIP_CALL_LEG_MODIFY_STATE_REINVITE_SENT
AppCallLegModifyResultRcvdEvHandler, StatusCode: 200
1-RVSIP_CALL_LEG_MODIFY_STATE_REINVITE_RESPONSE_RCVD
Sending RVSIP_METHOD_ACK1-RVSIP_CALL_LEG_MODIFY_STATE_IDLE
Selected codec t38fax
CID-0 T.38 EV_DETECT_FAX ....
1-RvSipCallLegDisconnect
Sending RVSIP_METHOD_ACK1-RVSIP_CALL_LEG_STATE_DISCONNECTING
1-RVSIP_CALL_LEG_STATE_DISCONNECTED
1-RVSIP_CALL_LEG_STATE_TERMINATED
********************************************



что конкретно не так в логах? что там - в логах, я имею ввиду - должно быть, чтобы факс нормально отправлялся?
2008-10-30 18:00

contramundum
Сообщений: 89

Re: t.38 Fax & * 1.4.18.1

dimas:

А чисто из любопытства - вы пробовали micronet напрямую с dlink без астериска соединить? Ходят ли там факсы? А то если верить логу, ваши стороны прекрасно договариваются друг с другом на T38 UDPTL _в обход_ астериска. Так что что там дальше между ними происходит науке неизвестно - 30 сек они занимаются каким-то сексом напрямую а потом уже жалуются на несудьбу...
)))))))
честно говоря, даже не представляю как можно напрямую соединить д-линк и микронет! В смысле, физически-то это не сложно, а вот логически, чтоб звонить-то можно было - хз...
2008-10-30 18:03

Evgen
Откуда: Саратов
Сообщений: 414

Re: t.38 Fax & * 1.4.18.1

В приведённых выше логах отбой шел от длинка (192.168.100.21)
<--- SIP read from 192.168.100.21:5060 --->
BYE sip:33@192.168.100.10 SIP/2.0
В логах микронета всё в порядке.
Сделайте всё-таки udptl debug во время попытки передачи факса
+7(925)140-7438
2008-10-30 18:21

cron333
Сообщений: 1573

Re: t.38 Fax & * 1.4.18.1

Показывайте sip.conf с настройками обоих пиров + global. Контекст диалплана, который обрабатывает этот вызов.

P.S. Попробуйте еще: t38pt_usertpsource=yes (хотя в вашей версии * этого параметра кажется еще не было)

P.P.S. Что показывает - CLI>udptl debug
2008-10-30 23:13

cron333
Сообщений: 1573

Re: t.38 Fax & * 1.4.18.1

С такими параметрами не пробовали?

t38pt_rtp=no
t38pt_tcp=no

Еще, в файле udptl.conf у вас похоже так:

;T38FaxUdpEC = t38UDPFEC
T38FaxUdpEC = t38UDPRedundancy

попробуйте сделать так:

T38FaxUdpEC = t38UDPFEC
;T38FaxUdpEC = t38UDPRedundancy



2008-10-31 01:03

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