chan_sip.c: Stopping retransmission on

Asterisk Asterisk Support
1

Hi,

I am using asterisk and SIPDiscount for PSTN termination. I however get about 95% termination failure and I guess the problem could be with my server configuration or an asterisk bug. I have attached my logs below and would be grateful for any useful tips.
I judge the “chan_sip.c: Stopping retransmission on …” could be a clue but I don’t know what conclusions to draw and what action to take.

Thanks in advance, Roland.

Dec 31 07:20:08 VERBOSE[5187] logger.c: Scheduling destruction of call 'C43A7DE9-5CB9-48CA-9BBA-044F7E820703@192.168.2.33' in 15000 ms
Dec 31 07:20:08 VERBOSE[5187] logger.c: Found user 'myusername'
Dec 31 07:20:08 VERBOSE[5187] logger.c: 
<-- SIP read from 80.134.90.108:5060: 
ACK sip:00492212597092@62.75.184.220 SIP/2.0
Via: SIP/2.0/UDP 80.134.90.108:5060;rport;branch=z9hG4bK1A682FBFDEA34DBDB48A165B4A6627F3
From: Roland Fru <sip:myusername@62.75.184.220>;tag=905104535
To: <sip:00492212597092@62.75.184.220>;tag=as228200f9
Contact: <sip:myusername@80.134.90.108:5060>
Call-ID: C43A7DE9-5CB9-48CA-9BBA-044F7E820703@192.168.2.33
CSeq: 46476 ACK
Max-Forwards: 70
Content-Length: 0


Dec 31 07:20:08 VERBOSE[5187] logger.c: --- (9 headers 0 lines)Dec 31 07:20:08 VERBOSE[5187] logger.c: --- (9 headers 0 lines)---
Dec 31 07:20:08 DEBUG[5187] chan_sip.c: Stopping retransmission on 'C43A7DE9-5CB9-48CA-9BBA-044F7E820703@192.168.2.33' of Response 46476: Match Found
Dec 31 07:20:08 VERBOSE[5187] logger.c: 
<-- SIP read from 80.134.90.108:5060: 
INVITE sip:00492212597092@62.75.184.220 SIP/2.0
Via: SIP/2.0/UDP 80.134.90.108:5060;rport;branch=z9hG4bK494EEC5E64ED4BD681F4C5BCAA106C5C
From: Roland Fru <sip:myusername@62.75.184.220>;tag=905104535
To: <sip:00492212597092@62.75.184.220>
Contact: <sip:myusername@80.134.90.108:5060>
Call-ID: C43A7DE9-5CB9-48CA-9BBA-044F7E820703@192.168.2.33
CSeq: 46477 INVITE
Proxy-Authorization: Digest username="myusername",realm="asterisk",nonce="52227048",response="3ae8c1c0703ab18e2b45528a46bdda96",uri="sip:00492212597092@62.75.184.220"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 235

v=0
o=myusername 3054803 3054860 IN IP4 80.134.90.108
s=X-Lite
c=IN IP4 80.134.90.108
t=0 0
m=audio 8000 RTP/AVP 3 98 101
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
2

Is that all there was in your debug? You should have received an ACK after your INVITE, did that ever happen?