Re: [asterisk-users] SIPp scenario file for testing UAC Authentication with Asterisk ?
On Thu, Oct 25, 2018 at 4:32 AM Olivier wrote: > Hello, > > I'm testing an Asterisk instance. > At the moment, I'm focusing on its capability to receive and challenge > incoming SIP Registrations. > > If all you want to do is test inbound registrations you can find an example SIPp scenario in the Asterisk testsuite[1]. You'll want to remove the section from the 200 response and the variable reference. Then you'll want to execute the scenario with something like the following (replacing with your values of course): sipp -m 1 -sf register-auth.xml -s -ap Another example can be found here[2]. Both the README and register.xml file have instructions on how to execute the test. Currently the test is setup to test a few thousand endpoints though. However you can adjust that number by modifying the register.csv (injection file) or by not using the injection file and modifying the register.xml scenario itself. [1] https://github.com/asterisk/testsuite/blob/master/tests/channels/pjsip/registration/inbound/nominal/single_contact/authed/sipp/register-auth.xml [2] http://blogs.asterisk.org/wp-content/uploads/2018/09/performance_inbound_registration.tar.gz > For various reasons, I would prefer to use SIPp instead of Asterisk to act > as SIP Client. > > Has someone successfully done this ? > If negative, what explains this ? > If positive, can you give an example of a successful SIPp scenario file ? > I've played with both embeded branchc and [1] but met no success yet > > Best regards > > [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml > > -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: https://digium.com & https://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to force Asterisk to reply with floating IP with chan_sip ?
Hello, I'm setting up a new cluster that must replace several old Asterisk instances. For various reasons, this new cluster must use chan_sip (migration to PJSIP is planned in a later phase). This new cluster uses VRRP in active/passive mode: - at any time, only one cluster member is active, - when a member becomes active, it inherits several floating IP addresses, it changes its IP configuration (with ip rule or ip route statements) and it starts Asterisk. Beside tweaking Linux IP configuration, is there a way to teach Asterisk's chan_sip "to always reply using as IP source, the destination IP it got the SIP request from" ? For instance, if Asterisk's chan_sip listen on IPs IP A, Ip B and IP C, all belonging to the same network, then I want Asterisk to reply with IP A for any request it received through IP A. Looking at ASTDB SIP/Registry instances, Asterisk does not save the IP address it heard an incoming REGISTER with. Did I miss something ? Suggestions ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip aor stays in status created
On Thu, Oct 25, 2018 at 6:58 AM marek cervenka wrote: > hi, > > i have webrtc client chrome69/jssip which is connecting to asterisk > 13.23.1/pjsip > > i have strange problem where pjsip aor stays in status "created" > > sip trace on asterisk looks ok. > > > do you think if this can be bug? > It is not a bug. The contact has been "created". It will stay in that state unless you are also going to qualify the endpoint. Asterisk 16 simply renames the state to "NonQualified" to be more explicit. Richard > > test*CLI> pjsip show aors > >Aor: > > Contact: > > > == > >Aor: vr1k50 1 > Contact: vr1k50/sip:6i2b9766@1.1.1.1:34434;tran b2ad914030 > Created 0.000 > > > > > <--- Received SIP request (566 bytes) from WSS:1.1.1.1:34434 ---> > REGISTER sip:sip.example.com SIP/2.0 > Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK2155317 > Max-Forwards: 69 > To: > From: "vr1k50" ;tag=d56ij3vuo3 > Call-ID: 0mm678kf72bc9b5ur7ea8d > CSeq: 13 REGISTER > Contact: > ;transport=ws>;+sip.ice;reg-id=1;+sip.instance="";expires=60 > Expires: 60 > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO > Supported: path,gruu,outbound > User-Agent: JsSIP 3.2.9 > Content-Length: 0 > > > <--- Transmitting SIP response (484 bytes) to WSS:1.1.1.1:34434 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/WSS > v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK2155317 > Call-ID: 0mm678kf72bc9b5ur7ea8d > From: "vr1k50" ;tag=d56ij3vuo3 > To: ;tag=z9hG4bK2155317 > CSeq: 13 REGISTER > WWW-Authenticate: Digest > > realm="asterisk",nonce="1540467808/121f72ae15612cc46a72e2861657a940",opaque="3060464337b28725",algorithm=md5,qop="auth" > Server: Asterisk PBX 13.23.1 > Content-Length: 0 > > > <--- Received SIP request (837 bytes) from WSS:1.1.1.1:34434 ---> > REGISTER sip:sip.example.com SIP/2.0 > Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK9799804 > Max-Forwards: 69 > To: > From: "vr1k50" ;tag=d56ij3vuo3 > Call-ID: 0mm678kf72bc9b5ur7ea8d > CSeq: 14 REGISTER > Authorization: Digest algorithm=MD5, username="vr1k50", > realm="asterisk", nonce="1540467808/121f72ae15612cc46a72e2861657a940", > uri="sip:sip.example.com", response="376b4ac58b01dde2e043931467bba55a", > opaque="3060464337b28725", qop=auth, cnonce="v8i7444gio8r", nc=0001 > Contact: > ;transport=ws>;+sip.ice;reg-id=1;+sip.instance="";expires=60 > Expires: 60 > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO > Supported: path,gruu,outbound > User-Agent: JsSIP 3.2.9 > Content-Length: 0 > > > <--- Transmitting SIP response (446 bytes) to WSS:1.1.1.1:34434 ---> > SIP/2.0 200 OK > Via: SIP/2.0/WSS > v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK9799804 > Call-ID: 0mm678kf72bc9b5ur7ea8d > From: "vr1k50" ;tag=d56ij3vuo3 > To: ;tag=z9hG4bK9799804 > CSeq: 14 REGISTER > Date: Thu, 25 Oct 2018 11:43:28 GMT > Contact: ;expires=59 > Expires: 60 > Server: Asterisk PBX 13.23.1 > Content-Length: 0 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip aor stays in status created
hi, i have webrtc client chrome69/jssip which is connecting to asterisk 13.23.1/pjsip i have strange problem where pjsip aor stays in status "created" sip trace on asterisk looks ok. do you think if this can be bug? test*CLI> pjsip show aors Aor: Contact: == Aor: vr1k50 1 Contact: vr1k50/sip:6i2b9766@1.1.1.1:34434;tran b2ad914030 Created 0.000 <--- Received SIP request (566 bytes) from WSS:1.1.1.1:34434 ---> REGISTER sip:sip.example.com SIP/2.0 Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK2155317 Max-Forwards: 69 To: From: "vr1k50" ;tag=d56ij3vuo3 Call-ID: 0mm678kf72bc9b5ur7ea8d CSeq: 13 REGISTER Contact: ;+sip.ice;reg-id=1;+sip.instance="";expires=60 Expires: 60 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: path,gruu,outbound User-Agent: JsSIP 3.2.9 Content-Length: 0 <--- Transmitting SIP response (484 bytes) to WSS:1.1.1.1:34434 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WSS v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK2155317 Call-ID: 0mm678kf72bc9b5ur7ea8d From: "vr1k50" ;tag=d56ij3vuo3 To: ;tag=z9hG4bK2155317 CSeq: 13 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1540467808/121f72ae15612cc46a72e2861657a940",opaque="3060464337b28725",algorithm=md5,qop="auth" Server: Asterisk PBX 13.23.1 Content-Length: 0 <--- Received SIP request (837 bytes) from WSS:1.1.1.1:34434 ---> REGISTER sip:sip.example.com SIP/2.0 Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK9799804 Max-Forwards: 69 To: From: "vr1k50" ;tag=d56ij3vuo3 Call-ID: 0mm678kf72bc9b5ur7ea8d CSeq: 14 REGISTER Authorization: Digest algorithm=MD5, username="vr1k50", realm="asterisk", nonce="1540467808/121f72ae15612cc46a72e2861657a940", uri="sip:sip.example.com", response="376b4ac58b01dde2e043931467bba55a", opaque="3060464337b28725", qop=auth, cnonce="v8i7444gio8r", nc=0001 Contact: ;+sip.ice;reg-id=1;+sip.instance="";expires=60 Expires: 60 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: path,gruu,outbound User-Agent: JsSIP 3.2.9 Content-Length: 0 <--- Transmitting SIP response (446 bytes) to WSS:1.1.1.1:34434 ---> SIP/2.0 200 OK Via: SIP/2.0/WSS v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK9799804 Call-ID: 0mm678kf72bc9b5ur7ea8d From: "vr1k50" ;tag=d56ij3vuo3 To: ;tag=z9hG4bK9799804 CSeq: 14 REGISTER Date: Thu, 25 Oct 2018 11:43:28 GMT Contact: ;expires=59 Expires: 60 Server: Asterisk PBX 13.23.1 Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I connect an existing Confbridge to a new SIP channel when DIALEDPEERNAME is empty?
Would really appreciate some help here - into day 4 of trying to bridge a
PJSIP call to an existing confbridge.
There's a fair amount of dialplan and log to show which doesn't really work
well via plain text email, so I've taken it over to the forum at
https://community.asterisk.org/t/bridging-an-existing-conference-to-a-new-call/76806/7
Many thanks in advance.
On Wed, 24 Oct 2018 at 17:17, Jonathan H wrote:
> Asterisk 16.0, PJSIP
>
> For the first caller to a conference, I want to dial out and bridge that
> conference to a new PJSIP external call.
>
> For the next callers, I just want them to join the local Asterisk
> conference.
>
> After the last caller leaves the conference, I want to hangup the call it
> initiated.
>
> Most of this works, but there are two problems - after the dial string and
> username is done sending, no further audio flows between the Confbridge
> conference and the external call.
>
> Secondly, I understand that I need the name of the "dialling out" channel:
>
> https://wiki.asterisk.org/wiki/display/AST/Pre-Bridge+Handlers
> > This application sets the following channel variables:
> > DIALEDPEERNAME - The name of the outbound channel that answered the call.
>
> But DIALEDPEERNAME is empty. Can anyone please suggest where I might be
> going wrong here, and how to complete this? Thank you!
>
> [bcab-dial-zoom]
> exten => s,1,Answer()
> same => n,Dial(PJSIP/0203456789@voipfone-201,,U(bcab-send-dtmf))
>
> [bcab-send-dtmf]
> exten => s,1,Wait(1)
> same => n,Verbose(1,***Dialled channel is ${DIALEDPEERNAME}); just
> gives :**Dialled channel is
> same => n,Set(dialedname=${DIALEDPEERNAME})
> same => n,SendDTMF(WW123456#W#W)
> same => n,Playback(technical-support)
> same => n,SendDTMF(#)
>
> same => n,SET(GOSUB_RESULT=GOTO:bcab-bridge-conference^s^1)
> same => n,Return()
>
> [bcab-bridge-conference]
> exten => s,1,Verbose(1,*** Entered bcab-bridge-conference)
> same => n,Answer()
> same => n,ConfBridge(1234)
> same => n,Wait(55)
> same => n,Hangup()
>
>
--
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Astricon is coming up October 9-11! Signup is available at:
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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[asterisk-users] SIPp scenario file for testing UAC Authentication with Asterisk ?
Hello, I'm testing an Asterisk instance. At the moment, I'm focusing on its capability to receive and challenge incoming SIP Registrations. For various reasons, I would prefer to use SIPp instead of Asterisk to act as SIP Client. Has someone successfully done this ? If negative, what explains this ? If positive, can you give an example of a successful SIPp scenario file ? I've played with both embeded branchc and [1] but met no success yet Best regards [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
