[ https://issues.asterisk.org/jira/browse/ASTERISK-26441?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232661#comment-232661 ]
Frederic Steinfels commented on ASTERISK-26441:
-----------------------------------------------
maybe this log might help:
maybe this log might help:
ha_ast1*CLI> core set verbose 1000
Console verbose is still 1000.
ha_ast1*CLI> core set debug 1000
Core debug was OFF and is now 1000.
ha_ast1*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:xx.xx.xx.xx:5060 --->
INVITE sip:s at yy.yy.yy.yy:5060 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKf3qpiv10581giu8fb4k0.1
Max-Forwards: 68
From: "07912345678" <sip:07912345678 at yy.yy.yy.yy:5069>;tag=as12de7e11
To: <sip:0430000000 at localproxy:5060>
Contact: <sip:07912345678 at xx.xx.xx.xx:5060;transport=udp>
Call-ID: 5018b1c43a30c0c915e360f527a4f66a at yy.yy.yy.yy:5069
CSeq: 102 INVITE
User-Agent: e-fon
Date: Wed, 12 Oct 2016 11:20:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-IPCONNECT: 0430000000
X-Number: 0430000000
P-Asserted-Identity: "07912345678" <sip:07912345678 at yy.yy.yy.yy>
Content-Type: application/sdp
Content-Length: 409
v=0
o=root 955680351 955680351 IN IP4 xx.xx.xx.xx
s=Asterisk PBX 11.21.1
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 13668 RTP/AVP 8 9 111 3 97 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (17 headers 18 lines) ---
Sending to xx.xx.xx.xx:5060 (no NAT)
Sending to xx.xx.xx.xx:5060 (no NAT)
Using INVITE request as basis request - 5018b1c43a30c0c915e360f527a4f66a at yy.yy.yy.yy:5069
Found peer '0435440806' for '07912345678' from xx.xx.xx.xx:5060
ha_ast1*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
> crash in ast_rtp_ice_add_cand
> -----------------------------
>
> Key: ASTERISK-26441
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-26441
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General, Resources/res_rtp_asterisk
> Affects Versions: 14.0.2
> Environment: Fedora 24, 64bit, i7-5820K
> Reporter: Frederic Steinfels
> Assignee: Unassigned
> Attachments: gdb.txt
>
>
> When I try to place a call from, asterisks immediately crashes. The crash always happens in ast_rtp_ice_add_cand. This behaviour is new. Asterisk was working flawlessly for years up to about saturday october 1. I have no clue what has happened on that day. I have moved my asterisk to another fedora 24 machine with less horsepower where it is working flawlessly with the same configuration files. I have compiled version 14.0.2 from source, I tried the original fedora 13.7.2 rpms and the fedora source 13.7.2 rpms. Same behaviour with all of them.
--
This message was sent by Atlassian JIRA
(v6.2#6252)