[asterisk-bugs] [JIRA] (ASTERISK-26337) Received "Forbidden" on qualify messages
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Wed Sep 7 16:38:01 CDT 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-26337?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-26337:
------------------------------------
Description:
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}
was:
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:
[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
With debug enabled for the peer "O2Line", the qualify communication looks like this:
*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
> 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
>
> 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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