Hello, I am using FreePBX for configuration (TrixBox 2.2). Asterisk 1.2.18 is connected to a VoIP provider using a SIP trunk. The provider is using Acmepaket /Metaswitch gear. Inbound and outbound calling work, but only to provisioned extensions. Inbound Routing (i.e. matching any/any CLID/DID) to a DISA/IVR or anything else for that matter does not work. Traces show a 404 is being sent back to the network. Here's example indicating the described behavior from Asterisk (sip debug enabled): INVITE sip:4002602@10.70.0.196:5060 SIP/2.0 Via: SIP/2.0/UDP 10.70.0.193:5060;branch=z9hG4bK0ngjtd202g7gfbsd47s1.1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Allow-Events: presence Max-Forwards: 69 Call-ID: SD8fofa01-0dfaf46e328a203a4ec2d7691d8d4ea2-v3000i1 From: ;isup-oli=00;tag=SD8fofa01-xxx.xxxx.com+1+35ac30+891f56b0 To: CSeq: 412230031 INVITE Expires: 180 Organization: Supported: 100rel Content-Length: 162 Content-Type: application/sdp Contact: ;isup-oli=00 P-Asserted-Identity: v=0 o=- 3934957908 3934957908 IN IP4 10.70.0.193 s=- c=IN IP4 10.70.0.193 t=0 0 m=audio 9434 RTP/AVP 2 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 Jun 2 00:42:08 VERBOSE[3794] logger.c: --- (19 headers 8 lines) --- Jun 2 00:42:08 VERBOSE[3794] logger.c: Using INVITE request as basis request - SD8fofa01-0dfaf46e328a203a4ec2d7691d8d4ea2-v3000i1 Jun 2 00:42:08 VERBOSE[3794] logger.c: Sending to 10.70.0.193 : 5060 (non-NAT) Jun 2 00:42:08 VERBOSE[3794] logger.c: Found peer 'QT' Jun 2 00:42:08 DEBUG[3794] chan_sip.c: Setting NAT on RTP to 524288 Jun 2 00:42:08 VERBOSE[3794] logger.c: Found RTP audio format 2 Jun 2 00:42:08 VERBOSE[3794] logger.c: Found RTP audio format 0 Jun 2 00:42:08 VERBOSE[3794] logger.c: Found RTP audio format 101 Jun 2 00:42:08 VERBOSE[3794] logger.c: Peer audio RTP is at port 10.70.0.193:9434 Jun 2 00:42:08 DEBUG[3794] chan_sip.c: Peer audio RTP is at port 10.70.0.193:9434 Jun 2 00:42:08 VERBOSE[3794] logger.c: Found description format telephone-event Jun 2 00:42:08 VERBOSE[3794] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jun 2 00:42:08 VERBOSE[3794] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 2 00:42:08 DEBUG[3794] chan_sip.c: Checking SIP call limits for device 4414002600 Jun 2 00:42:08 VERBOSE[3794] logger.c: Looking for 4002602 in default (domain 10.70.0.196) Jun 2 00:42:08 VERBOSE[3794] logger.c: Reliably Transmitting (NAT) to 10.70.0.193:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.70.0.193:5060;branch=z9hG4bK0ngjtd202g7gfbsd47s1.1;received=10.70.0.193 From: ;isup-oli=00;tag=SD8fofa01-xxx.xxxx.com+1+35ac30+891f56b0 To: ;tag=as54797227 Call-ID: SD8fofa01-0dfaf46e328a203a4ec2d7691d8d4ea2-v3000i1 CSeq: 412230031 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Jun 2 00:42:08 VERBOSE[3794] logger.c: <-- SIP read from 10.70.0.193:5060: ACK sip:4002602@10.70.0.196:5060 SIP/2.0 Via: SIP/2.0/UDP 10.70.0.193:5060;branch=z9hG4bK0ngjtd202g7gfbsd47s1.1 CSeq: 412230031 ACK Max-Forwards: 69 Call-ID: SD8fofa01-0dfaf46e328a203a4ec2d7691d8d4ea2-v3000i1 From: ;isup-oli=00;tag=SD8fofa01-xxx.xxxx.com+1+35ac30+891f56b0 To: ;tag=as54797227 Content-Length: 0 I have packet captures off the network and sip debug if anyone interested. Grateful for any suggestions, Cheers, Mike Mike Harris ONXIS 26 Victoria Street, Hamilton, HMCX, Bermuda. O: 441-400-4001 M: 441-505-7153 MORE DEBUG (As Requested by TheCompWiz) <-- SIP read from 10.70.0.193:5060: INVITE sip:4002602@10.70.0.196:5060 SIP/2.0 Via: SIP/2.0/UDP 10.70.0.193:5060;branch=z9hG4bKgebg893070k10b0pb3g1.1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Allow-Events: presence Max-Forwards: 69 Call-ID: SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 From: ;isup-oli=00;tag=SD8vrba01-quantum1.quantum.bm+1+25053a+f171d803 To: CSeq: 155893236 INVITE Expires: 180 Organization: Supported: 100rel Content-Length: 162 Content-Type: application/sdp Contact: ;isup-oli=00 P-Asserted-Identity: v=0 o=- 4165324218 4165324218 IN IP4 10.70.0.193 s=- c=IN IP4 10.70.0.193 t=0 0 m=audio 9550 RTP/AVP 2 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 0: INVITE sip:4002602@10.70.0.196:5060 SIP/2.0 (43) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.70.0.193:5060;branch=z9hG4bKgebg893070k10b0pb3g1.1 (70) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 2: Allow-Events: message-summary (29) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 3: Allow-Events: refer (19) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 4: Allow-Events: dialog (20) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 5: Allow-Events: line-seize (24) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 6: Allow-Events: presence (22) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 7: Max-Forwards: 69 (16) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 8: Call-ID: SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 (59) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 9: From: ;isup-oli=00;tag=SD8vrba01-quantum1.quantum.bm+1+25053a+f171d803 (117) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 10: To: (29) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 11: CSeq: 155893236 INVITE (22) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 12: Expires: 180 (12) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 13: Organization: (14) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 14: Supported: 100rel (17) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 15: Content-Length: 162 (19) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 16: Content-Type: application/sdp (29) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 17: Contact: ;isup-oli=00 (68) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 18: P-Asserted-Identity: (52) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 19: (0) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Line: v=0 (3) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Line: o=- 4165324218 4165324218 IN IP4 10.70.0.193 (44) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Line: s=- (3) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Line: c=IN IP4 10.70.0.193 (20) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Line: t=0 0 (5) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Line: m=audio 9550 RTP/AVP 2 0 101 (28) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Line: a=ptime:20 (10) Jun 4 16:41:32 VERBOSE[3794] logger.c: --- (19 headers 8 lines) --- Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Allocating new SIP dialog for SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 - INVITE (With RTP) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Begin: parsing SIP "Supported: 100rel" Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Found SIP option: -100rel- Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Matched SIP option: 100rel Jun 4 16:41:32 DEBUG[3794] chan_sip.c: * SIP extension value: 2 for call SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 Jun 4 16:41:32 VERBOSE[3794] logger.c: Using INVITE request as basis request - SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 Jun 4 16:41:32 VERBOSE[3794] logger.c: Sending to 10.70.0.193 : 5060 (non-NAT) Jun 4 16:41:32 VERBOSE[3794] logger.c: Found peer 'Quantum' Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Setting NAT on RTP to 524288 Jun 4 16:41:32 VERBOSE[3794] logger.c: Found RTP audio format 2 Jun 4 16:41:32 VERBOSE[3794] logger.c: Found RTP audio format 0 Jun 4 16:41:32 VERBOSE[3794] logger.c: Found RTP audio format 101 Jun 4 16:41:32 VERBOSE[3794] logger.c: Peer audio RTP is at port 10.70.0.193:9550 Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Peer audio RTP is at port 10.70.0.193:9550 Jun 4 16:41:32 VERBOSE[3794] logger.c: Found description format telephone-event Jun 4 16:41:32 VERBOSE[3794] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jun 4 16:41:32 VERBOSE[3794] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Checking SIP call limits for device 4414002600 Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Updating call counter for incoming call Jun 4 16:41:32 VERBOSE[3794] logger.c: Looking for 4002602 in default (domain 10.70.0.196) Jun 4 16:41:32 VERBOSE[3794] logger.c: Reliably Transmitting (NAT) to 10.70.0.193:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.70.0.193:5060;branch=z9hG4bKgebg893070k10b0pb3g1.1;received=10.70.0.193 From: ;isup-oli=00;tag=SD8vrba01-quantum1.quantum.bm+1+25053a+f171d803 To: ;tag=as67a0e37e Call-ID: SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 CSeq: 155893236 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Jun 4 16:41:32 DEBUG[3794] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #58436 Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Updating call counter for incoming call Jun 4 16:41:32 VERBOSE[3794] logger.c: <-- SIP read from 10.70.0.193:5060: ACK sip:4002602@10.70.0.196:5060 SIP/2.0 Via: SIP/2.0/UDP 10.70.0.193:5060;branch=z9hG4bKgebg893070k10b0pb3g1.1 CSeq: 155893236 ACK Max-Forwards: 69 Call-ID: SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 From: ;isup-oli=00;tag=SD8vrba01-quantum1.quantum.bm+1+25053a+f171d803 To: ;tag=as67a0e37e Content-Length: 0 Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 0: ACK sip:4002602@10.70.0.196:5060 SIP/2.0 (40) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.70.0.193:5060;branch=z9hG4bKgebg893070k10b0pb3g1.1 (70) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 2: CSeq: 155893236 ACK (19) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 3: Max-Forwards: 69 (16) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 4: Call-ID: SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 (59) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 5: From: ;isup-oli=00;tag=SD8vrba01-quantum1.quantum.bm+1+25053a+f171d803 (117) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 6: To: ;tag=as67a0e37e (44) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 7: Content-Length: 0 (17) Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Header 8: (0) Jun 4 16:41:32 VERBOSE[3794] logger.c: --- (8 headers 0 lines) --- Jun 4 16:41:32 DEBUG[3794] chan_sip.c: = Found Their Call ID: SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1 Their Tag SD8vrba01-quantum1.quantum.bm+1+25053a+f171d803 Our tag: as67a0e37e Jun 4 16:41:32 DEBUG[3794] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 4 16:41:32 DEBUG[3794] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #58436 Jun 4 16:41:32 DEBUG[3794] chan_sip.c: Stopping retransmission on 'SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1' of Response 155893236: Match Found Jun 4 16:41:32 VERBOSE[3794] logger.c: Destroying call 'SD8vrba01-167f24b1e4eedc72e25858555aa759d4-v3000i1' Jun 4 16:41:42 DEBUG[509] manager.c: Manager received command 'login' Jun 4 16:41:42 VERBOSE[509] logger.c: == Parsing '/etc/asterisk/manager.conf': Jun 4 16:41:42 DEBUG[509] config.c: Parsing /etc/asterisk/manager.conf Jun 4 16:41:42 VERBOSE[509] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Jun 4 16:41:42 VERBOSE[509] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Jun 4 16:41:42 DEBUG[509] config.c: Parsing /etc/asterisk/manager_custom.conf Jun 4 16:41:42 VERBOSE[509] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Found Jun 4 16:41:42 DEBUG[509] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer Jun 4 16:41:42 DEBUG[509] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer Jun 4 16:41:42 DEBUG[509] acl.c: ##### Testing 127.0.0.1 with 0.0.0.0 Jun 4 16:41:42 DEBUG[509] acl.c: ##### Testing 127.0.0.1 with 127.0.0.0 Jun 4 16:41:42 VERBOSE[509] logger.c: == Manager 'admin' logged on from 127.0.0.1 Jun 4 16:41:42 VERBOSE[509] logger.c: == Manager 'admin' logged off from 127.0.0.1 Jun 4 16:41:42 VERBOSE[509] logger.c: == Manager 'admin' logged off from 127.0.0.1