Лог 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" ;tag=as5aa7c089 To: Contact: 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: From: "606498" ;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: ;tag=a8befbcff3b57e9i1 From: "606498" ;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 ;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: ;tag=a8befbcff3b57e9i1 From: "606498" ;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 Server: Linksys/SPA8000-5.1.10 Remote-Party-ID: 606025 ;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: set_destination: Parsing 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" ;tag=as5aa7c089 To: ;tag=a8befbcff3b57e9i1 Contact: 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 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" ;tag=as5aa7c089 To: ;tag=a8befbcff3b57e9i1 Contact: 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: ;tag=a8befbcff3b57e9i1 From: "606498" ;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 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 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" ;tag=as5aa7c089 To: ;tag=a8befbcff3b57e9i1 Contact: 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 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" ;tag=as5aa7c089 To: ;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: ;tag=a8befbcff3b57e9i1 From: "606498" ;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