############### zapata.conf ##################### [trunkgroups] [channels] callprogress=no usecallerid=yes hidecallerid=no callwaiting=no echocancel=yes echotraining=yes context=incoming signalling=fxs_ks channel => 4 ################################################# ############ zaptel.conf ######################## fxsks=4 loadzone=us defaultzone=us ################################################## ####### extensions.conf ######################### [general] static=yes writeprotect=no priorityjumping=no [globals] [default] exten => _0XXXXXXXX,1,Dial(Zap/4/${EXTEN},40) exten => _0XXXXXXXX,2,Congestion() ################################################# ##### asterisk -rvvvvvvvvvv with sip debug ####### Connected to Asterisk 1.2.13 currently running on caerj (pid = 21707) Verbosity was 0 and is now 10 caerj*CLI> sip debug SIP Debugging enabled caerj*CLI> <-- SIP read from 10.10.3.190:5060: INVITE sip:027528182@caerj.proderj.rj.gov.br SIP/2.0 Record-Route: Date: Mon, 23 Jul 2007 19:57:20 GMT CSeq: 1 INVITE Via: SIP/2.0/UDP 10.10.3.190;branch=z9hG4bK086b7097-c437-dc11-991b-0013d44f0ad0 Via: SIP/2.0/UDP 10.10.2.33:5072;branch=z9hG4bK086b7097-c437-dc11-991b-0013d44f0ad0;rport=5072 User-Agent: Ekiga/2.0.3 From: "paulo arruda" ;tag=5c6e6e97-c437-dc11-991b-0013d44f0ad0 Call-ID: 26686e97-c437-dc11-991b-0013d44f0ad0@gsl024 To: Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 383 Max-Forwards: 69 v=0 o=- 1185220640 1185220640 IN IP4 10.10.2.33 s=Opal SIP Session c=IN IP4 10.10.2.33 t=0 0 m=audio 5036 RTP/AVP 101 96 3 107 110 0 8 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:96 SPEEX/16000 a=rtpmap:3 GSM/8000 a=rtpmap:107 MS-GSM/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 m=video 5038 RTP/AVP 31 a=rtpmap:31 H261/90000 --- (15 headers 16 lines) --- Using INVITE request as basis request - 26686e97-c437-dc11-991b-0013d44f0ad0@gsl024 Sending to 10.10.3.190 : 5060 (non-NAT) Jul 23 16:53:58 WARNING[21722]: res_config_odbc.c:88 realtime_odbc: SQL Alloc Handle failed! Jul 23 16:53:58 WARNING[21722]: res_config_odbc.c:88 realtime_odbc: SQL Alloc Handle failed! Jul 23 16:53:58 WARNING[21722]: res_config_odbc.c:88 realtime_odbc: SQL Alloc Handle failed! Found no matching peer or user for '10.10.3.190:5060' Found RTP audio format 101 Found RTP audio format 96 Found RTP audio format 3 Found RTP audio format 107 Found RTP audio format 110 Found RTP audio format 0 Found RTP audio format 8 Jul 23 16:53:58 WARNING[21722]: chan_sip.c:3602 process_sdp: Unknown SDP media type in offer: video 5038 RTP/AVP 31 Peer audio RTP is at port 10.10.2.33:5036 Found description format telephone-event Found description format SPEEX Found description format GSM Found description format MS-GSM Found description format SPEEX Found description format PCMU Found description format PCMA Found description format H261 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 027528182 in default (domain caerj.proderj.rj.gov.br) list_route: hop: Transmitting (no NAT) to 10.10.3.190:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.3.190;branch=z9hG4bK086b7097-c437-dc11-991b-0013d44f0ad0;received=10.10.3.190 Via: SIP/2.0/UDP 10.10.2.33:5072;branch=z9hG4bK086b7097-c437-dc11-991b-0013d44f0ad0;rport=5072 From: "paulo arruda" ;tag=5c6e6e97-c437-dc11-991b-0013d44f0ad0 To: Call-ID: 26686e97-c437-dc11-991b-0013d44f0ad0@gsl024 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 23 16:53:58 WARNING[21711]: res_config_odbc.c:88 realtime_odbc: SQL Alloc Handle failed! -- Executing Dial("SIP/caerj.proderj.rj.gov.br-08189e10", "Zap/4/027528182|40") in new stack -- Called 4/027528182 -- Zap/4-1 answered SIP/caerj.proderj.rj.gov.br-08189e10 We're at 10.10.3.190 port 16326 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.10.3.190:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.3.190;branch=z9hG4bK086b7097-c437-dc11-991b-0013d44f0ad0;received=10.10.3.190 Via: SIP/2.0/UDP 10.10.2.33:5072;branch=z9hG4bK086b7097-c437-dc11-991b-0013d44f0ad0;rport=5072 Record-Route: From: "paulo arruda" ;tag=5c6e6e97-c437-dc11-991b-0013d44f0ad0 To: ;tag=as216f4415 Call-ID: 26686e97-c437-dc11-991b-0013d44f0ad0@gsl024 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 21707 21707 IN IP4 10.10.3.190 s=session c=IN IP4 10.10.3.190 t=0 0 m=audio 16326 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 23 16:54:02 WARNING[21711]: res_config_odbc.c:88 realtime_odbc: SQL Alloc Handle failed! caerj*CLI> <-- SIP read from 10.10.3.190:5060: ACK sip:027528182@10.10.3.190:5070 SIP/2.0 Record-Route: CSeq: 1 ACK Via: SIP/2.0/UDP 10.10.3.190;branch=z9hG4bK68c4dc99-c437-dc11-991b-0013d44f0ad0 Via: SIP/2.0/UDP 10.10.2.33:5072;branch=z9hG4bK68c4dc99-c437-dc11-991b-0013d44f0ad0;rport=5072 From: "paulo arruda" ;tag=5c6e6e97-c437-dc11-991b-0013d44f0ad0 Call-ID: 26686e97-c437-dc11-991b-0013d44f0ad0@gsl024 To: ;tag=as216f4415 Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Length: 0 Max-Forwards: 69 P-hint: rr-enforced --- (13 headers 0 lines) --- caerj*CLI>