Hi, I have a problem with a SIP trunk between Asterisk and central OXE Alcatel, especially sometimes are not received inbound calls with following messages:
-- Executing [...@test:1] AGI("SIP/800-084250f8", "agi://127.0.0.1/test.agi") in new stack -- AGI Script agi://127.0.0.1/test.agi completed, returning 0 == Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN' I configured the sip.conf file: [800] type=peer host=172.XX.XX.XX username=test secret=XXXXXXX insecure=very context=test disallow=all allow=alaw allow=ulaw and the extensions.conf file: exten => 375,1,AGI(agi://127.0.0.1/test.agi) I attach to this email the sip messages receveid by Asterisk when the problem occurs. Thanks for your help. Best regards, GP
<--- SIP read from 172.25.51.1:10011 ---> INVITE sip:3...@172.24.10.188;user=phone SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: "ISDN_T2" <sip:+521776...@mercury> Content-Type: application/sdp To: <sip:3...@172.24.10.188;user=phone> From: "ISDN_T2" <sip:+521776...@mercury>;tag=cc01ff37a60521d35da001f98edda0ac Contact: sip:172.25.51.1 Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 CSeq: 684819861 INVITE Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c Max-Forwards: 70 Content-Length: 314 v=0 o=OXE 1279704517 1279704517 IN IP4 172.25.51.1 s=abs c=IN IP4 172.25.51.4 t=0 0 m=audio 32712 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 <-------------> --- (13 headers 17 lines) --- Sending to 172.25.51.1 : 5060 (no NAT) Using INVITE request as basis request - 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 Found peer '800' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 172.25.51.4:32712 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.25.51.4:32712 Looking for 375 in sedoc (domain 172.24.10.188) list_route: hop: <sip:172.25.51.1> <--- Transmitting (no NAT) to 172.25.51.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1 From: "ISDN_T2" <sip:+521776...@mercury>;tag=cc01ff37a60521d35da001f98edda0ac To: <sip:3...@172.24.10.188;user=phone> Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 CSeq: 684819861 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:3...@172.24.10.188> Content-Length: 0 <------------> -- Executing [...@sedoc:1] AGI("SIP/800-084250f8", "agi://127.0.0.1/mercury.agi") in new stack -- AGI Script agi://127.0.0.1/mercury.agi completed, returning 0 == Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN' Scheduling destruction of SIP dialog '82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 172.25.51.1:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1 From: "ISDN_T2" <sip:+521776...@mercury>;tag=cc01ff37a60521d35da001f98edda0ac To: <sip:3...@172.24.10.188;user=phone>;tag=as3455cb36 Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 CSeq: 684819861 INVITE ser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> ccsedoc*CLI> <--- SIP read from 172.25.51.1:10011 ---> ACK sip:3...@172.24.10.188;user=phone SIP/2.0 Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 From: "ISDN_T2" <sip:+521776...@mercury>;tag=cc01ff37a60521d35da001f98edda0ac To: <sip:3...@172.24.10.188;user=phone>;tag=as3455cb36 Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c CSeq: 684819861 ACK Content-Length: 0 <------------->
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