[asterisk-users] VoIP Information in CDRs

[Mindaugas Kezys]

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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Costa Tsaousis
Sent: 2009 m. vasario 21 d. 17:30
To: asterisk-users at 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 [h at 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
'0aa4f73f5c9715b7661b50080a66901f at 10.11.12.1' in 6656 ms (Method: INVITE)
set_destination: Parsing  <sip:422 at 10.11.12.43:5060;transport=udp>
<sip:422 at 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:422 at 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:401 at 10.11.12.1>
<sip:401 at 10.11.12.1>;tag=as1d9352fe
To:  <sip:422 at 10.11.12.43:5060;transport=udp>
<sip:422 at 10.11.12.43:5060;transport=udp>;tag=0009b7aa1aaa51eb2c767e13-7fb3b3
4a
Call-ID: 0aa4f73f5c9715b7661b50080a66901f at 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:401 at 10.11.12.1>
<sip:401 at 10.11.12.1>;tag=as1d9352fe
To:  <sip:422 at 10.11.12.43:5060;transport=udp>
<sip:422 at 10.11.12.43:5060;transport=udp>;tag=0009b7aa1aaa51eb2c767e13-7fb3b3
4a
Call-ID: 0aa4f73f5c9715b7661b50080a66901f at 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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