Voila les logs lorsque j'effectue un appel (j'ai volontairement masqué les adresses IP et numéros de téléphone) :
Code:
-- Executing [s@ivr-recording:4] Record("SIP/xx.xx.xx.xx-0000006d", "/var/lib/asterisk/sounds/records/recorded/test_20150914_213018:wav,0,1800,k") in new stack
-- <SIP/xx.xx.xx.xx-0000006d> Playing 'beep.gsm' (language 'fr')
Lorsque ca coupe :
Code:
<--- SIP read from UDP:xx.xx.xx.xx:5070 --->
BYE sip:+33XXXXXXXXX@xx.xx.xx.xx:5070 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5070;rport;branch=z9hG4bK19HSm38Qy39Uj
Max-Forwards: 70
From: "+33XXXXXXXXX" <sip:+33XXXXXXXXX@xx.xx.xx.xx>;tag=rm99rHH7FB81N
To: <sip:+33XXXXXXXXX@xxx.com>;tag=as33909228
Call-ID: 36d1-4c7-814201519-fsc1.gwin
CSeq: 80793471 BYE
Contact: <sip:mod_sofia@xx.xx.xx.xx:5070>
User-Agent: Sewan_TRUNKFSC1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Supported: path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to xx.xx.xx.xx:5070 (NAT)
Scheduling destruction of SIP dialog '36d1-4c7-814201519-fsc1.gwin' in 32000 ms (Method: BYE)
<--- Transmitting (NAT) to xx.xx.xx.xx:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx:5070;branch=z9hG4bK19HSm38Qy39Uj;received=xx.xx.xx.xx;rport=5070
From: "+33XXXXXXXXX" <sip:+33XXXXXXXXX@xx.xx.xx.xx>;tag=rm99rHH7FB81N
To: <sip:+33XXXXXXXXX@xxx.com>;tag=as33909228
Call-ID: 36d1-4c7-814201519-fsc1.gwin
CSeq: 80793471 BYE
Server: Asterisk PBX 13.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Voila la variable RECORD_STATUS :
Code:
-- Executing [h@ivr-recording:1] Verbose("SIP/xx.xx.xx.xx-0000006f", "1,recordstatus=HANGUP") in new stack
recordstatus=HANGUP
A noter que j'ai dans la CLI d'asterisk, souvent ce genre de message qui s'affiche, cela a t il un rapport :
Code:
---
Retransmitting #X (NAT) to xx.xx.xx.xx:5070:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xx.xx.xx.xx:5070;branch=z9hG4bK-1b270b3175c078e6eb086c3300abac68;received=xx.xx.xx.xx;rport=5070
From: 8966<sip:8966@xx.xx.xx.xx>;tag=6f2acc6d
To: 00972598549491<sip:00972598549491@xx.xx.xx.xx>;tag=as56148f85
Call-ID: 1b270b3175c078e6eb086c3300abac68
CSeq: 1 INVITE
Server: Asterisk PBX 13.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
J'en ai 10 (Retransmitting #X ou X va de 1 Ã 10) et ensuite j'ai ce message :
Code:
[Sep 14 21:37:58] WARNING[28098]: chan_sip.c:3996 retrans_pkt: Retransmission timeout reached on transmission 77afa89e2e7513b99b8491287c91b5cb for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Really destroying SIP dialog '77afa89e2e7513b99b8491287c91b5cb' Method: INVITE
Pour info ma version d'asterisk est 13.3