I'm trying to find the answer to the same question:
4. Find out who hangedup an answered call. HANGUPCAUSE = 16 DIALSTATUS = ANSWERER In both cases, so these variables does not help. Can anybody help with this issue? Should be pretty simple to detect which part hanguped the call first. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Costa Tsaousis Sent: 2009 m. vasario 21 d. 17:30 To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoIP Information in CDRs Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)} Codec=${CHANNEL(audioreadformat)}/${CHANNEL(audiowriteformat)}/${CHANNEL(aud ionativeformat)}/${SIPCHANINFO(t38passthrough)} QOS=${RTPAUDIOQOS}) The problems I have so far: 1. CODEC Codec is reported only for A-Leg. When transcoding asterisk logs the above line as: slin for read / slin for write / the codec of A-Leg / 0 for t.38. Is there a way to get the codec for both legs of a call? 2. RTP Qos is reported only for A-Leg. Also, asterisk seems to ignore the RTP statistics reports by B-Leg after the BYE: -- Executing [...@core-dialplan:3] Hangup("SIP/401-08231540", "") in new stack == Spawn h extension (core-dialplan, h, 3) exited non-zero on 'SIP/401-08231540' Scheduling destruction of SIP dialog '0aa4f73f5c9715b7661b50080a669...@10.11.12.1' in 6656 ms (Method: INVITE) set_destination: Parsing <sip:4...@10.11.12.43:5060;transport=udp> <sip:4...@10.11.12.43:5060;transport=udp> for address/port to send to set_destination: set destination to 10.11.12.43, port 5060 Reliably Transmitting (no NAT) to 10.11.12.43:5060: BYE sip:4...@10.11.12.43:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.11.12.1:5060;branch=z9hG4bK3077b703;rport From: "Office Line 1" <sip:4...@10.11.12.1> <sip:4...@10.11.12.1>;tag=as1d9352fe To: <sip:4...@10.11.12.43:5060;transport=udp> <sip:4...@10.11.12.43:5060;transport=udp>;tag=0009b7aa1aaa51eb2c767e13-7fb3b3 4a Call-ID: 0aa4f73f5c9715b7661b50080a669...@10.11.12.1 CSeq: 103 BYE User-Agent: home.tsaousis.gr Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (core-dialplan, 422, 1) exited non-zero on 'SIP/401-08231540' box*CLI> <--- SIP read from 10.11.12.43:50539 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.12.1:5060;branch=z9hG4bK3077b703;rport From: "Office Line 1" <sip:4...@10.11.12.1> <sip:4...@10.11.12.1>;tag=as1d9352fe To: <sip:4...@10.11.12.43:5060;transport=udp> <sip:4...@10.11.12.43:5060;transport=udp>;tag=0009b7aa1aaa51eb2c767e13-7fb3b3 4a Call-ID: 0aa4f73f5c9715b7661b50080a669...@10.11.12.1 Date: Sat, 21 Feb 2009 14:29:42 GMT CSeq: 103 BYE Server: Cisco-CP7960G/8.0 Content-Length: 0 RTP-RxStat: Dur=4,Pkt=180,Oct=28800,LatePkt=0,LostPkt=0,AvgJit=0 RTP-TxStat: Dur=4,Pkt=183,Oct=29280 These SIP messages are being exchanged after the dialplan has executed the h extension. Is there a way to have RTP statistics for both legs? 3. RTP IP is not reported anywhere. The RIP= variable I have above, reports the SIP IP, and again only for A-Leg. Is it possible to find out the RTP (not SIP) IPs for both legs? 4. Find out who hangedup an answered call. I have not found any way to determine the peer that requested to hangup the call. Is it possible to find who of the two legs requested the hangup? Any help is appreciated. Costa
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