[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:11194 handle_response_invite: SIP response 407 to standard invite
Transmitting (NAT) to 217.114.78.114:5060:
ACK sip:01716344354@sip.pepphone.de SIP/2.0
Via: SIP/2.0/UDP 88.198.230.110:5060;branch=z9hG4bK450f0efe;rport
From: "507430" <sip:507430@sip.pepphone.de>;tag=as083c447c
To: <sip:01716344354@sip.pepphone.de>;tag=as1a5bd0ab
Contact: <sip:507430@88.198.230.110>
Call-ID: 7749805236360fcc2f2e838d3f336f49@sip.pepphone.de
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:10620 do_proxy_auth: Auth attempt 1 on INVITE
[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:5820 add_sdp: ** Our capability: 0x2 (gsm) Video flag: False
[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:5821 add_sdp: ** Our prefcodec: 0x2 (gsm)
Audio is at 88.198.230.110 port 12072
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:5973 add_sdp: -- Done with adding codecs to SDP
[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:5974 add_sdp: XXX p->owner=='0'
Segmentation fault (core dumped)