t38 gateway

Необходимо, чтобы заработал t38 gateway в Asterisk. Использую Asterisk 1.8.10 вот с этим патчем https://issues.asterisk.org/view.php?id=13405.

Схема включения:

FAX1 <> FXS <> ASTERISK1 <> IP <> ASTERISK2 <> FXS <> FAX2

Между двумя астерисками sip trunk. Настройки факса в sip.conf

t38pt_udptl = yes,redundancy(пробовал все возможные). 

Экстеншн на входящие:

[fax] exten => 118,1,NoOp() exten => 118,n,Set(FAXOPT(gateway)=yes) exten => 118,n,Dial(SLIC//dev/si3226x_fxs1.0,600) 

Экстеншн на исходящие:

[fxs1] exten => _7.,1,NoOp() exten => _7.,n,Set(FAXOPT(gateway)=yes) exten => _7.,n,Dial(SIP/trunk_2/${EXTEN:1}, 30) 

Астериски с двух сторон настраиваю симметрично.

Вот часть вывода asterisk -rd:

<-------------> [Jun 14 12:30:54] VERBOSE[3309] chan_sip.c: --- (13 headers 0 lines) --- [Jun 14 12:30:54] VERBOSE[3309] chan_sip.c: Scheduling destruction of SIP dialog '2cb2500a4dbe2f3a74ef218c33233d73@192.168.10.88' in 32000 ms (Method: REGISTER) [Jun 14 12:30:54] NOTICE[3309] chan_sip.c: Outbound Registration: Expiry for 192.168.10.88 is 120 sec (Scheduling reregistration in 105 s) [Jun 14 12:31:17] DEBUG[3311] config.c: Parsing /etc/asterisk/si3226x.conf [Jun 14 12:31:17] VERBOSE[3311] chan_si3226x.c: -- event_offhook, iface: /dev/si3226x_fxs1.0 [Jun 14 12:31:17] VERBOSE[3311] chan_si3226x.c: -- AST_STATE_DOWN: [Jun 14 12:31:19] NOTICE[3311] chan_si3226x.c: SLIC_GET_DTMF_ASCII: 55 [Jun 14 12:31:19] VERBOSE[3311] chan_si3226x.c: -- event_dtmf 7 [Jun 14 12:31:19] NOTICE[3311] chan_si3226x.c: state: 2 [Jun 14 12:31:20] NOTICE[3311] chan_si3226x.c: SLIC_GET_DTMF_ASCII: 49 [Jun 14 12:31:20] VERBOSE[3311] chan_si3226x.c: -- event_dtmf 1 [Jun 14 12:31:20] NOTICE[3311] chan_si3226x.c: state: 3 [Jun 14 12:31:20] NOTICE[3311] chan_si3226x.c: SLIC_GET_DTMF_ASCII: 49 [Jun 14 12:31:20] VERBOSE[3311] chan_si3226x.c: -- event_dtmf 1 [Jun 14 12:31:20] NOTICE[3311] chan_si3226x.c: state: 3 [Jun 14 12:31:21] NOTICE[3311] chan_si3226x.c: SLIC_GET_DTMF_ASCII: 56 [Jun 14 12:31:21] VERBOSE[3311] chan_si3226x.c: -- event_dtmf 8 [Jun 14 12:31:21] NOTICE[3311] chan_si3226x.c: state: 3 [Jun 14 12:31:23] VERBOSE[3311] chan_si3226x.c: -- event_digit_timer [Jun 14 12:31:23] VERBOSE[3311] chan_si3226x.c: -- extension exists, starting PBX 7118 [Jun 14 12:31:23] VERBOSE[3311] chan_si3226x.c: -- Sarting audio transfer on line /dev/si3226x_fxs1.0 [Jun 14 12:31:23] DEBUG[3315] pbx.c: Launching 'NoOp' [Jun 14 12:31:23] DEBUG[3315] pbx.c: Launching 'Set' [Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jun 14 12:31:23] DEBUG[3315] pbx.c: Launching 'Dial' [Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Allocating new SIP dialog for 68ce19ef4f2231bd42aa84640213ea44@(null) - INVITE (No RTP) [Jun 14 12:31:23] DEBUG[3315] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x12df758' [Jun 14 12:31:23] DEBUG[3315] res_rtp_asterisk.c: Allocated port 15234 for RTP instance '0x12df758' [Jun 14 12:31:23] DEBUG[3315] rtp_engine.c: RTP instance '0x12df758' is setup and ready to go [Jun 14 12:31:23] DEBUG[3315] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x12df758' [Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Setting NAT on RTP to Off [Jun 14 12:31:23] DEBUG[3315] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jun 14 12:31:23] DEBUG[3315] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable DIALEDTIME. [Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable ANSWEREDTIME. [Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable DIALEDPEERNAME. [Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable DIALEDPEERNUMBER. [Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable DIALSTATUS. [Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Outgoing Call for 118 [Jun 14 12:31:23] DEBUG[3315] chan_sip.c: ** Our capability: 0x80e (gsm|ulaw|alaw|g726) Video flag: False Text flag: False [Jun 14 12:31:23] DEBUG[3315] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Audio is at 15234 [Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding codec 0x800 (g726) to SDP [Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Initializing initreq for method INVITE - callid 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.88:5060: INVITE sip:118@192.168.10.88 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK36358623 Max-Forwards: 70 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88> Contact: <sip:118@192.168.10.89:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.10.0 Date: Thu, 14 Jun 2012 12:31:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 342 v=0 o=root 862485137 862485137 IN IP4 192.168.10.89 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.89 t=0 0 m=audio 15234 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK36358623;received=192.168.10.89 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 102 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:118@192.168.10.88:5060> Content-Length: 0 <-------------> [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: --- (11 headers 0 lines) --- [Jun 14 12:31:24] DEBUG[3309] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' Request 102: Found [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK36358623;received=192.168.10.89 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 102 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:118@192.168.10.88:5060> Content-Type: application/sdp Content-Length: 344 v=0 o=root 1997296106 1997296106 IN IP4 192.168.10.88 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.88 t=0 0 m=audio 19418 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: --- (12 headers 15 lines) --- [Jun 14 12:31:24] DEBUG[3309] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' Request 102: Found [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: list_route: hop: <sip:118@192.168.10.88:5060> [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 0 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 0 based on m type on 0x4090f0e8 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 8 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 8 based on m type on 0x4090f0e8 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 3 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 3 based on m type on 0x4090f0e8 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 111 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 111 based on m type on 0x4090f0e8 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 101 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 101 based on m type on 0x4090f0e8 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format GSM for ID 3 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format G726-32 for ID 111 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 0 on 0x4090f0e8 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 3 on 0x4090f0e8 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 8 on 0x4090f0e8 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 101 on 0x4090f0e8 [Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 111 on 0x4090f0e8 [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x80e (gsm|ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x80e (gsm|ulaw|alaw|g726) [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 14 12:31:24] DEBUG[3309] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x12df758' [Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Peer audio RTP is at port 192.168.10.88:19418 [Jun 14 12:31:24] DEBUG[3309] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x12df758' [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK36358623;received=192.168.10.89 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 102 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:118@192.168.10.88:5060> Content-Type: application/sdp Content-Length: 344 v=0 o=root 1997296106 1997296107 IN IP4 192.168.10.88 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.88 t=0 0 m=audio 19418 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: --- (12 headers 15 lines) --- [Jun 14 12:31:26] DEBUG[3309] chan_sip.c: Acked pending invite 102 [Jun 14 12:31:26] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Request 102: Match Found [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 0 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 0 based on m type on 0x4090f0e8 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 8 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 8 based on m type on 0x4090f0e8 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 3 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 3 based on m type on 0x4090f0e8 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 111 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 111 based on m type on 0x4090f0e8 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 101 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 101 based on m type on 0x4090f0e8 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format GSM for ID 3 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format G726-32 for ID 111 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 0 on 0x4090f0e8 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 3 on 0x4090f0e8 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 8 on 0x4090f0e8 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 101 on 0x4090f0e8 [Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 111 on 0x4090f0e8 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x80e (gsm|ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x80e (gsm|ulaw|alaw|g726) [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 14 12:31:26] DEBUG[3309] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x12df758' [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Peer audio RTP is at port 192.168.10.88:19418 [Jun 14 12:31:26] DEBUG[3309] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x12df758' [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: list_route: hop: <sip:118@192.168.10.88:5060> [Jun 14 12:31:26] DEBUG[3309] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: set_destination: set destination to 192.168.10.88:5060 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Transmitting (no NAT) to 192.168.10.88:5060: ACK sip:118@192.168.10.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK55c36014 Max-Forwards: 70 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Contact: <sip:118@192.168.10.89:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.10.0 Content-Length: 0 --- [Jun 14 12:31:26] DEBUG[3315] features.c: Removing dialed interfaces datastore on SIP/trunk_2-00000000 since we're bridging [Jun 14 12:31:26] DEBUG[3315] channel.c: Set channel /dev/si3226x_fxs1.0 to read format slin [Jun 14 12:31:26] DEBUG[3315] channel.c: Set channel /dev/si3226x_fxs1.0 to write format slin [Jun 14 12:31:26] DEBUG[3315] channel.c: Set channel SIP/trunk_2-00000000 to read format slin [Jun 14 12:31:26] DEBUG[3315] channel.c: Set channel SIP/trunk_2-00000000 to write format slin [Jun 14 12:31:26] VERBOSE[3315] chan_si3226x.c: -- start audio transfer, set silence state on line /dev/si3226x_fxs1.0 [Jun 14 12:31:26] DEBUG[3315] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Jun 14 12:31:26] DEBUG[3315] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Really destroying SIP dialog '2cb2500a4dbe2f3a74ef218c33233d73@192.168.10.88' Method: REGISTER [Jun 14 12:31:31] DEBUG[3315] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Jun 14 12:31:36] DEBUG[3315] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Jun 14 12:31:38] DEBUG[3315] dsp.c: 2100 Hz done detected [Jun 14 12:31:38] NOTICE[3315] channel.c: Dropping incompatible voice frame on SIP/trunk_2-00000000 of format slin since our native format has changed to 0x4 (ulaw) [Jun 14 12:31:38] DEBUG[3315] chan_sip.c: Setting NAT on UDPTL to Off [Jun 14 12:31:38] DEBUG[3315] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:38] VERBOSE[3315] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to [Jun 14 12:31:38] VERBOSE[3315] chan_sip.c: set_destination: set destination to 192.168.10.88:5060 [Jun 14 12:31:38] DEBUG[3315] chan_sip.c: T.38 UDPTL is at 192.168.10.89 port 4551 [Jun 14 12:31:38] DEBUG[3315] chan_sip.c: Initializing already initialized SIP dialog 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 (presumably reinvite) [Jun 14 12:31:38] VERBOSE[3315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.88:5060: INVITE sip:118@192.168.10.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK17a5c3ac Max-Forwards: 70 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Contact: <sip:118@192.168.10.89:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 862485137 862485138 IN IP4 192.168.10.89 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.89 t=0 0 m=image 4551 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC --- [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> INVITE sip:118@192.168.10.89:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK2ad630a1 Max-Forwards: 70 From: <sip:118@192.168.10.88>;tag=as0bead541 To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b Contact: <sip:118@192.168.10.88:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 267 v=0 o=root 1997296106 1997296108 IN IP4 192.168.10.88 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.88 t=0 0 m=image 4541 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC <-------------> [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: --- (14 headers 11 lines) --- [Jun 14 12:31:38] WARNING[3309] chan_sip.c: Failed to read an alternate host or port in SDP. Expect audio problems [Jun 14 12:31:38] WARNING[3309] chan_sip.c: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call. [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.88:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK2ad630a1;received=192.168.10.88 From: <sip:118@192.168.10.88>;tag=as0bead541 To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 102 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <------------> [Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Got INVITE on call where we already have pending INVITE, deferring that - 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK17a5c3ac;received=192.168.10.89 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 103 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: --- (12 headers 0 lines) --- [Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Acked pending invite 103 [Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Request 103: Match Found [Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: set_destination: set destination to 192.168.10.88:5060 [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: Transmitting (no NAT) to 192.168.10.88:5060: ACK sip:118@192.168.10.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK17a5c3ac Max-Forwards: 70 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Contact: <sip:118@192.168.10.89:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.10.0 Content-Length: 0 --- [Jun 14 12:31:38] WARNING[3309] chan_sip.c: just did sched_add waitid(13) for sip_reinvite_retry for dialog 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 in handle_response_invite [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> ACK sip:118@192.168.10.89:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK2ad630a1 Max-Forwards: 70 From: <sip:118@192.168.10.88>;tag=as0bead541 To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b Contact: <sip:118@192.168.10.88:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.10.0 Content-Length: 0 <-------------> [Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: --- (10 headers 0 lines) --- [Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Response 102: Match Found [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> INVITE sip:118@192.168.10.89:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK21f9b33c Max-Forwards: 70 From: <sip:118@192.168.10.88>;tag=as0bead541 To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b Contact: <sip:118@192.168.10.88:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 267 v=0 o=root 1997296106 1997296109 IN IP4 192.168.10.88 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.88 t=0 0 m=image 4541 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC <-------------> [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: --- (14 headers 11 lines) --- [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Sending to 192.168.10.88:5060 (no NAT) [Jun 14 12:31:40] DEBUG[3309] chan_sip.c: Initializing initreq for method INVITE - callid 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Got T.38 offer in SDP in dialog 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. [Jun 14 12:31:40] DEBUG[3309] chan_sip.c: Peer T.38 UDPTL is at port 192.168.10.88:4541 [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: <--- Transmitting (no NAT) to 192.168.10.88:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK21f9b33c;received=192.168.10.88 From: <sip:118@192.168.10.88>;tag=as0bead541 To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 103 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:118@192.168.10.89:5060> Content-Length: 0 <------------> [Jun 14 12:31:40] DEBUG[3315] chan_sip.c: T.38 UDPTL is at 192.168.10.89 port 4551 [Jun 14 12:31:40] VERBOSE[3315] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.10.88:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK21f9b33c;received=192.168.10.88 From: <sip:118@192.168.10.88>;tag=as0bead541 To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 103 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:118@192.168.10.89:5060> Content-Type: application/sdp Content-Length: 265 v=0 o=root 862485137 862485139 IN IP4 192.168.10.89 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.89 t=0 0 m=image 4551 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC <------------> [Jun 14 12:31:40] DEBUG[3315] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> ACK sip:118@192.168.10.89:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK78a876b1 Max-Forwards: 70 From: <sip:118@192.168.10.88>;tag=as0bead541 To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b Contact: <sip:118@192.168.10.88:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.10.0 Content-Length: 0 <-------------> [Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: --- (10 headers 0 lines) --- [Jun 14 12:31:40] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Response 103: Match Found [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: set_destination: set destination to 192.168.10.88:5060 [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: ** Our capability: 0x80e (gsm|ulaw|alaw|g726) Video flag: True Text flag: True [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Audio is at 15234 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding codec 0x800 (g726) to SDP [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Initializing already initialized SIP dialog 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 (presumably reinvite) [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.88:5060: INVITE sip:118@192.168.10.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK656f8b61 Max-Forwards: 70 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Contact: <sip:118@192.168.10.89:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 342 v=0 o=root 862485137 862485140 IN IP4 192.168.10.89 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.89 t=0 0 m=audio 15234 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK656f8b61;received=192.168.10.89 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 104 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:118@192.168.10.88:5060> Content-Length: 0 <-------------> [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: --- (11 headers 0 lines) --- [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' Request 104: Found [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: <--- SIP read from UDP:192.168.10.88:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK656f8b61;received=192.168.10.89 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 104 INVITE Server: Asterisk PBX 1.8.10.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:118@192.168.10.88:5060> Content-Type: application/sdp Content-Length: 344 v=0 o=root 1997296106 1997296110 IN IP4 192.168.10.88 s=Asterisk PBX 1.8.10.0 c=IN IP4 192.168.10.88 t=0 0 m=audio 19418 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: --- (12 headers 15 lines) --- [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Acked pending invite 104 [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Request 104: Match Found [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 0 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 0 based on m type on 0x4090f0e8 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 8 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 8 based on m type on 0x4090f0e8 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 3 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 3 based on m type on 0x4090f0e8 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 111 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 111 based on m type on 0x4090f0e8 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 101 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 101 based on m type on 0x4090f0e8 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format GSM for ID 3 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format G726-32 for ID 111 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 0 on 0x4090f0e8 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 3 on 0x4090f0e8 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 8 on 0x4090f0e8 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 101 on 0x4090f0e8 [Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 111 on 0x4090f0e8 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x80e (gsm|ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x80e (gsm|ulaw|alaw|g726) [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Peer audio RTP is at port 192.168.10.88:19418 [Jun 14 12:31:41] DEBUG[3309] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x12df758' [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Peer doesn't provide T.38 UDPTL [Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: set_destination: set destination to 192.168.10.88:5060 [Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Transmitting (no NAT) to 192.168.10.88:5060: ACK sip:118@192.168.10.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK286706cd Max-Forwards: 70 From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b To: <sip:118@192.168.10.88>;tag=as0bead541 Contact: <sip:118@192.168.10.89:5060> Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.10.0 Content-Length: 0 --- [Jun 14 12:31:41] DEBUG[3315] channel.c: Got a FRAME_CONTROL (31) frame on channel SIP/trunk_2-00000000 [Jun 14 12:31:41] DEBUG[3315] channel.c: Bridge stops bridging channels /dev/si3226x_fxs1.0 and SIP/trunk_2-00000000 [Jun 14 12:31:41] DEBUG[3315] res_fax.c: Incompatible T.38 State for gateway mode [Jun 14 12:31:41] DEBUG[3315] channel.c: Generator got voice, switching to phase locked mode [Jun 14 12:31:41] DEBUG[3315] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jun 14 12:31:41] DEBUG[3315] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x12df758' [Jun 14 12:31:41] DEBUG[3315] res_fax.c: Incompatible T.38 State for gateway mode [Jun 14 12:31:41] DEBUG[3315] res_fax.c: Incompatible T.38 State for gateway mode [Jun 14 12:31:41] DEBUG[3315] res_fax.c: Incompatible T.38 State for gateway mode 

Дальше долго идёт T.38 State for gateway mode. Есть ещё вывод wiresharkа, но он в текстовом виде как то плохо читаемый. Есть у кого какие мысли, почему факсы не проходят?