rendered paste bodyЛог попытки (неудачной) отправки факса с 606498 на 606025.
Не в строке ли (***) дело? Вместо NSE пробовал RE-Invite,
не помогло. Есть соображения?
bee*CLI> set verbose 3
Verbosity was 0 and is now 3
The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead.
-- Executing [606026@from.cisco:1] Dial("SIP/10.200.1.3-08331dd8", "SIP/606026||40") in new stack
-- Called 606026
-- Got SIP response 486 "Busy Here" back from 10.200.0.34
-- SIP/606026-082b5250 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [606026@from.cisco:2] Congestion("SIP/10.200.1.3-08331dd8", "") in new stack
== Spawn extension (from.cisco, 606026, 2) exited non-zero on 'SIP/10.200.1.3-08331dd8'
bee*CLI> sip set debug peer 606025
SIP Debugging Enabled for IP: 10.200.0.34:5061
== Spawn extension (from.sip, 84955134203, 3) exited non-zero on 'SIP/606026-081fdc80'
-- Executing [84957109907@from.sip:1] NoOp("SIP/606018-081eac40", "") in new stack
-- Executing [84957109907@from.sip:2] Set("SIP/606018-081eac40", "RINGTONE=waitdial") in new stack
-- Executing [84957109907@from.sip:3] Dial("SIP/606018-081eac40", "SIP/84957109907@cisco||40") in new stack
-- Called 84957109907@cisco
-- SIP/cisco-08237f98 is making progress passing it to SIP/606018-081eac40
-- SIP/cisco-08237f98 answered SIP/606018-081eac40
-- Packet2Packet bridging SIP/606018-081eac40 and SIP/cisco-08237f98
-- Executing [606026@from.cisco:1] Dial("SIP/10.200.1.3-08253a40", "SIP/606026||40") in new stack
-- Called 606026
-- Got SIP response 486 "Busy Here" back from 10.200.0.34
-- SIP/606026-08254fd0 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [606026@from.cisco:2] Congestion("SIP/10.200.1.3-08253a40", "") in new stack
== Spawn extension (from.cisco, 606026, 2) exited non-zero on 'SIP/10.200.1.3-08253a40'
-- Executing [84955134203@from.sip:1] NoOp("SIP/606026-08226c28", "") in new stack
-- Executing [84955134203@from.sip:2] Set("SIP/606026-08226c28", "RINGTONE=waitdial") in new stack
-- Executing [84955134203@from.sip:3] Dial("SIP/606026-08226c28", "SIP/84955134203@cisco||40") in new stack
-- Called 84955134203@cisco
-- SIP/cisco-0835c748 is making progress passing it to SIP/606026-08226c28
-- Executing [606025@from.sip8:1] Dial("SIP/606498-08236a08", "SIP/606025||40") in new stack
Audio is at 10.200.0.33 port 10070
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.200.0.34:5061:
INVITE sip:606025@10.1.16.34:5061 SIP/2.0
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK2a27d614;rport
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
To: <sip:606025@10.1.16.34:5061>
Contact: <sip:606498@10.200.0.33>
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 10 Apr 2008 09:53:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 30202 30202 IN IP4 10.200.0.33
s=session
c=IN IP4 10.200.0.33
t=0 0
m=audio 10070 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:40
a=sendrecv
---
-- Called 606025
bee*CLI>
<--- SIP read from 10.200.0.34:5061 --->
SIP/2.0 100 Trying
To: <sip:606025@10.1.16.34:5061>
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK2a27d614
Server: Linksys/SPA8000-5.1.10
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
bee*CLI>
<--- SIP read from 10.200.0.34:5061 --->
SIP/2.0 180 Ringing
To: <sip:606025@10.1.16.34:5061>;tag=a8befbcff3b57e9i1
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK2a27d614
Server: Linksys/SPA8000-5.1.10
Remote-Party-ID: 606025 <sip:606025@10.200.0.33>;screen=yes;party=called
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/606025-08231810 is ringing
bee*CLI>
<--- SIP read from 10.200.0.34:5061 --->
SIP/2.0 200 OK
To: <sip:606025@10.1.16.34:5061>;tag=a8befbcff3b57e9i1
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK2a27d614
Contact: 606025 <sip:606025@10.1.16.34:5061>
Server: Linksys/SPA8000-5.1.10
Remote-Party-ID: 606025 <sip:606025@10.200.0.33>;screen=yes;party=called
Content-Length: 279
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 127993581 127993581 IN IP4 10.1.16.34
s=-
c=IN IP4 10.1.16.34
t=0 0
m=audio 16424 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (13 headers 14 lines) ---
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 10.1.16.34:16424
Found audio description format G729a for ID 18
*** Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.16.34:16424
list_route: hop: <sip:606025@10.1.16.34:5061>
set_destination: Parsing <sip:606025@10.1.16.34:5061> for address/port to send to
set_destination: set destination to 10.1.16.34, port 5061
Transmitting (NAT) to 10.200.0.34:5061:
ACK sip:606025@10.1.16.34:5061 SIP/2.0
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK4a5c3164;rport
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
To: <sip:606025@10.1.16.34:5061>;tag=a8befbcff3b57e9i1
Contact: <sip:606498@10.200.0.33>
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/606025-08231810 answered SIP/606498-08236a08
set_destination: Parsing <sip:606025@10.1.16.34:5061> for address/port to send to
set_destination: set destination to 10.1.16.34, port 5061
Reliably Transmitting (NAT) to 10.200.0.34:5061:
INVITE sip:606025@10.1.16.34:5061 SIP/2.0
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK086557be;rport
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
To: <sip:606025@10.1.16.34:5061>;tag=a8befbcff3b57e9i1
Contact: <sip:606498@10.200.0.33>
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
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 (T38 switchover)
Content-Type: application/sdp
Content-Length: 347
v=0
o=root 30202 30203 IN IP4 10.200.0.33
s=session
c=IN IP4 10.200.0.33
t=0 0
m=image 4958 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy
---
bee*CLI>
<--- SIP read from 10.200.0.34:5061 --->
SIP/2.0 200 OK
To: <sip:606025@10.1.16.34:5061>;tag=a8befbcff3b57e9i1
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 103 INVITE
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK086557be
Contact: 606025 <sip:606025@10.1.16.34:5061>
Server: Linksys/SPA8000-5.1.10
Content-Length: 267
Content-Type: application/sdp
v=0
o=- 127994032 127994032 IN IP4 10.1.16.34
s=-
c=IN IP4 10.1.16.34
t=0 0
m=image 16424 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (10 headers 12 lines) ---
Got T.38 offer in SDP in dialog 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:606025@10.1.16.34:5061> for address/port to send to
set_destination: set destination to 10.1.16.34, port 5061
Transmitting (NAT) to 10.200.0.34:5061:
ACK sip:606025@10.1.16.34:5061 SIP/2.0
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK22e51363;rport
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
To: <sip:606025@10.1.16.34:5061>;tag=a8befbcff3b57e9i1
Contact: <sip:606498@10.200.0.33>
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- Got SIP response 486 "Busy here" back from 10.200.1.3
-- SIP/cisco-0835c748 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [84955134203@from.sip:4] Congestion("SIP/606026-08226c28", "") in new stack
== Spawn extension (from.sip, 84955134203, 4) exited non-zero on 'SIP/606026-08226c28'
-- Executing [211@from.office:1] Dial("SIP/209-08331dd8", "SIP/211||40") in new stack
-- Called 211
-- SIP/211-081ec1d0 is ringing
-- SIP/211-081ec1d0 answered SIP/209-08331dd8
-- Packet2Packet bridging SIP/209-08331dd8 and SIP/211-081ec1d0
== Spawn extension (from.office, 211, 1) exited non-zero on 'SIP/209-08331dd8'
Scheduling destruction of SIP dialog '7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:606025@10.1.16.34:5061> for address/port to send to
set_destination: set destination to 10.1.16.34, port 5061
Reliably Transmitting (NAT) to 10.200.0.34:5061:
BYE sip:606025@10.1.16.34:5061 SIP/2.0
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK26ebbe94;rport
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
To: <sip:606025@10.1.16.34:5061>;tag=a8befbcff3b57e9i1
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (from.sip8, 606025, 1) exited non-zero on 'SIP/606498-08236a08'
bee*CLI>
<--- SIP read from 10.200.0.34:5061 --->
SIP/2.0 200 OK
To: <sip:606025@10.1.16.34:5061>;tag=a8befbcff3b57e9i1
From: "606498" <sip:606498@10.200.0.33>;tag=as5aa7c089
Call-ID: 7e21d69b525e82fc41a9a5376cc82a28@10.200.0.33
CSeq: 104 BYE
Via: SIP/2.0/UDP 10.200.0.33:5060;branch=z9hG4bK26ebbe94
Server: Linksys/SPA8000-5.1.10
Content-Length: 0