[asterisk-bugs] [JIRA] (ASTERISK-26337) Received "Forbidden" on qualify messages

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26337?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton reassigned ASTERISK-26337:
---------------------------------------

    Assignee: Rusty Newton

> Received "Forbidden" on qualify messages
> ----------------------------------------
>
>                 Key: ASTERISK-26337
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26337
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.13.1
>         Environment: Debian 8.5
>            Reporter: Hannes Kroeger
>            Assignee: Rusty Newton
>
> If "qualify" is set to "yes", the qualify messages are answered by "Forbidden" by the provider. I suspect, this is because the "fromdomain" setting is not properly respected in the qualify messages.
> The whole configuration only works for some minutes. After that, no incoming or outgoing calls can be made any more.
> The sip.conf section is here:
> {noformat}
> [O2Line]
> type=peer
> insecure=port,invite
> nat=force_rport
> username=493811216473
> fromuser=493811216473
> fromdomain=sip.alice-voip.de
> secret=***********
> host=sip.alice-voip.de
> qualify=no
> canreinvite=no
> dtmfmode=rfc2833
> context=default
> callbackextension=493811216473
> outboundproxy=sip.alice-voip.de
> trustrpid=yes
> sendrpid=no
> disallow=g729
> directmedia=no
> rtpkeepalive=60
> keepalive=yes
> {noformat}
> With debug enabled for the peer "O2Line", the qualify communication looks like this:
> {noformat}
> *CLI> sip qualify peer O2Line
> Reliably Transmitting (NAT) to 62.52.148.214:5060:
> OPTIONS sip:sip.alice-voip.de SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK4e8837c0;rport
> Max-Forwards: 70
> From: "asterisk" <sip:493811216473 at 192.168.1.7>;tag=as33536d28
> To: <sip:sip.alice-voip.de>
> Contact: <sip:493811216473 at 192.168.1.7:5060>
> Call-ID: 0e0cbec4112196de7781d9e706e727ed at 192.168.1.7:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
> Date: Mon, 05 Sep 2016 12:03:49 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
> ---
> <--- SIP read from UDP:62.52.148.214:5060 --->
> SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP 192.168.1.7:5060;received=77.186.233.123;branch=z9hG4bK4e8837c0;rport=5060
> From: "asterisk" <sip:493811216473 at 192.168.1.7:5060>;tag=as33536d28
> To: <sip:sip.alice-voip.de>;tag=aprqngfrt-ctpjjf30000c6
> Call-ID: 0e0cbec4112196de7781d9e706e727ed at 192.168.1.7:5060
> CSeq: 102 OPTIONS
> <------------->
> --- (6 headers 0 lines) ---
> Really destroying SIP dialog '0e0cbec4112196de7781d9e706e727ed at 192.168.1.7:5060' Method: OPTIONS
> {noformat}



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