[asterisk-bugs] [JIRA] (ASTERISK-25295) res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h

[Dmitriy Serov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25295?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227389#comment-227389 ] 

Dmitriy Serov commented on ASTERISK-25295:
------------------------------------------

bq. Does the contact uri for the following case actually contain the "=" at the end after "X-PUSH-URI"?
Yes. I don't have REGISTER packet dump (user is offline), but "sip show peer 16923" (now is chan_sip) displays:
{noformat}
  Reg. Contact : sip:16923 at 87.228.126.146:61276;app-id=s.notify.live.net;pn-type=wp;pn-tok=/u/1/hk2/H2QAAADXKbJW5qha12tAYboqAjIAKWDWJ4KaBWmMhMLehaBrOeFgFqEFfsl18pRPQ-cGpykPK-N0PszqaXcfZzqJLqFnEqHFSg-NhfE0cOmS6KSTq7UxS41Q90fLyQxeWsI_Fog/d2lu
{noformat}

in logs i found such packet:
{noformat}
REGISTER sip:talk37.ru;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:50985;rport;branch=z9hG4bKPjmQXgOIC3W.
Max-Forwards: 70
From: "ivmaster" <sip:ivmaster at talk37.ru>;tag=a0ZwQBBkHDataga30
To: "ivmaster" <sip:ivmaster at talk37.ru>
Call-ID: dAe1tzVT1G
CSeq: 18018 REGISTER
User-Agent: Join 3.2.4(10958)
Contact: "ivmaster" <sip:ivmaster at 192.168.1.66:50985>;reg-id=1;+sip.instance="<urn:uuid:C427A9B9-ECA3-8383-4AE6-BA41B88A180F>"
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, MESSAGE
Content-Length:  0
{noformat}

The users of LinphoneWP8KeepAlive/ (belle-sip/1.3.2) makes asterisk (chan_sip) to generate such packet:
{noformat}
OPTIONS sip:bondima at 109.188.124.22;app-id=s.notify.live.net;pn-type=wp;pn-tok=/u/1/sin/H2QAAABQox0UYTJcaRqfCbZudHMkHygU_K-GcGBrFvf89_n6chf18lP4JeQHUKX-5yPDO49eNoA260qgDtBfGFAIXvnrJVLJ_G_7uzbBTGUlQHSGnImJFsAn-i3KYxyp8pF6LHg/d2luZG93c3Bob25lZGVmYXVsdA/ZhRmmVyMm0ilZ SIP/2.0
Via: SIP/2.0/UDP 85.142.148.80:5060;branch=z9hG4bK5eff2789;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at talk37.ru>;tag=as249c56dc
To: <sip:bondima at 109.188.124.22;app-id=s.notify.live.net;pn-type=wp;pn-tok=/u/1/sin/H2QAAABQox0UYTJcaRqfCbZudHMkHygU_K-GcGBrFvf89_n6chf18lP4JeQHUKX-5yPDO49eNoA260qgDtBfGFAIXvnrJVLJ_G_7uzbBTGUlQHSGnImJFsAn-i3KYxyp8pF6LHg/d2luZG93c3Bob25lZGVmYXVsdA/ZhRmmVyMm0il
Contact: <sip:asterisk at 85.142.148.80:5060>
Call-ID: 61d6e12e28403b324a4af91e5a697de2 at talk37.ru
CSeq: 102 OPTIONS
User-Agent: ruVoIP.net PBX
Date: Fri, 28 Aug 2015 10:35:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
{noformat}

Users of winphone quite rare. I turned on sip debug and waiting "REGISTER" packet from.

Questions:
1. What version of pjproject better now to use?
2. What configure params to use?
I could make another attempt to switch devices on PJSIP with debug enabled and a full dump of the packets.

> res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-25295
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25295
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 13.4.0
>            Reporter: Dmitriy Serov
>            Assignee: Kevin Harwell
>         Attachments: 2015_08_10__20_38_07.backtrace-threads.txt, 2015_08_10__20_38_07.full.tail.txt, 2015_08_10__20_58_07.backtrace-threads.txt, 2015_08_10__20_58_07.full.tail.txt, backtrace.2015-07-30-1.txt, backtrace.2015-07-30-2.txt, backtrace.2015-07-30-3.txt, backtrace.2015-07-30-5.txt, core.back-trace.txt, debuglog.txt, full.2015-07-30-1.txt, full.2015-07-30-2.txt, full.2015-07-30-3.txt, full.2015-07-30-5.txt
>
>
> Using last git branch 13. Now is 13.5-rc.
> Tired of fighting with deadlock when using chan_sip. Decided to convert all the devices and the gateways to res_pjsip.
> The result was crash very often. When asterisk boots. Sometimes asterisk successfully loaded and crash in a few minutes.
> In the configs, nothing has changed except transfer chan_sip devices to res_pjsip.



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