Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?
"Joshua C. Colp" writes: >> I am curious if the "reuse registration TCP connection" is required by >> standards or if it is merely obviously good practice. >> >> I have had this problem too with asterisk 16.5.0 >> >> This is not the first recommendation I have seen to use kamailio as a >> proxy for asterisk, for these sorts of issues as well as clients that >> change addresses. Unfortunately the "jsr pc, set_up_kamailo" subroutine >> call is still executing so I can't say if things work right then... >> > > There is a specification for doing it, but it's not required by the main > SIP RFC. In fact the main one states that you're supposed to establish an > outgoing connection to the address in the Contact header. In practice, > though, this is futile as generally NAT Is in use so you can't connect back > and thus you reuse the connection. The chan_sip module should do this > automatically, while chan_pjsip will do this if "rewrite_contact" is set to > yes. Thanks. I read the docs for pjsip (and the book) and failed to grasp that. It seems that this should be the default behavior. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?
On 12/27/2019 2:56 PM, Jonathan H wrote: OK, that works - looks like a documentation bug? (Also very confusing!) Should I report it on the page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+AGICommand_get+full+variable or on the main tracker? These wiki pages are automatically generated from the source code[1], so the best place would be the Asterisk Issues Tracker[2]. Even better would be to attach a patch to the issue you create which beefs up the documentation to your liking. Kind regards, Sean [1] https://github.com/asterisk/asterisk/blob/17/res/res_agi.c#L321-L341 [2] https://issues.asterisk.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?
OK, that works - looks like a documentation bug? (Also very confusing!)
Should I report it on the page at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+AGICommand_get+full+variable
or
on the main tracker?
Many thanks - couldn't figure out what was wrong!
On Fri, 27 Dec 2019 at 19:36, Sean Bright wrote:
> On 12/27/2019 2:24 PM, Jonathan H wrote:
> > AGI Rx << SET VARIABLE myVar "Hello
> > World!!!"
> > AGI Tx >> 200 result=1
> > AGI Rx << GET FULL VARIABLE myVar
> > AGI Tx >> 200 result=1 (myVar)
> >
> > Is this a bug, poor documentation, or my poor understanding of them?
>
> I believe the syntax you are looking for is:
>
> GET FULL VARIABLE ${myVar}
>
> Kind regards,
> Sean
>
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Re: [asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?
On 12/27/2019 2:24 PM, Jonathan H wrote:
AGI Rx << SET VARIABLE myVar "Hello
World!!!"
AGI Tx >> 200 result=1
AGI Rx << GET FULL VARIABLE myVar
AGI Tx >> 200 result=1 (myVar)
Is this a bug, poor documentation, or my poor understanding of them?
I believe the syntax you are looking for is:
GET FULL VARIABLE ${myVar}
Kind regards,
Sean
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[asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?
Just trying out a node agi package (https://github.com/sergey12313/ts-agi/ , and it wasn't behaving as I expected, but when turning on agi debug, it looks like it might be Asterisk (using 17.1.0) This works as expected AGI Rx << SET VARIABLE myVar "Hello World!!!" AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE myVar AGI Tx >> 200 result=1 (Hello World!!!) But GET FULL VARIABLE just returns the NAME of the variable AGI Rx << SET VARIABLE myVar "Hello World!!!" AGI Tx >> 200 result=1 AGI Rx << GET FULL VARIABLE myVar AGI Tx >> 200 result=1 (myVar) Is this a bug, poor documentation, or my poor understanding of them? https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+AGICommand_get+full+variable GET FULL VARIABLE VARIABLENAME CHANNEL NAME Example return code: 200 result=1 (testvariable) https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+AGICommand_get+variable GET VARIABLE VARIABLENAME Example return code: 200 result=1 (testvariable) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] SIP via TCP - new TCP session per call or use same session for multiple calls?
On Fri, Dec 27, 2019 at 2:00 PM Greg Troxel wrote: > Dovid Bender writes: > > > So long as the tcp socket is open your SBC should send the call back over > > the same socket. Now it can be that your SBC is seeing the socket as > > timing out. If you are using Kamailio you can have it send tcp keep > alives > > every so often so that the socket stays up. > > SBC? > > I am curious if the "reuse registration TCP connection" is required by > standards or if it is merely obviously good practice. > > I have had this problem too with asterisk 16.5.0 > > This is not the first recommendation I have seen to use kamailio as a > proxy for asterisk, for these sorts of issues as well as clients that > change addresses. Unfortunately the "jsr pc, set_up_kamailo" subroutine > call is still executing so I can't say if things work right then... > There is a specification for doing it, but it's not required by the main SIP RFC. In fact the main one states that you're supposed to establish an outgoing connection to the address in the Contact header. In practice, though, this is futile as generally NAT Is in use so you can't connect back and thus you reuse the connection. The chan_sip module should do this automatically, while chan_pjsip will do this if "rewrite_contact" is set to yes. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Handling a non-responsive peer after it answers
On Fri, Dec 27, 2019 at 12:02 PM David P wrote:
>
> I'm looking for a way of detecting in my dialplan when a peer becomes
> non-responsive after answering. It seems that Asterisk knows when the peer
> becomes non-responsive because it logs "Remote UNIX connection
> disconnected" around the same time, and it seems that
> if there is no following "Remote UNIX connection" within a short time,
> then the peer can be considered non-responsive. Is there a way to configure
> a handler for this state?
>
> We use v14.7.6 and we dial the peer this way:
>
> same =>
> n,Set(CHANNEL(hangup_handler_push)=${CONTEXT},handleHangupByCaller,1(args))
> same =>
> n,Dial(${AddressToReachPeer},2,b(${CONTEXT}^afterDialingPeerLogIpOfCb^1(${UUID}^${StartEpoch})))
> same => n,Goto(handle${DIALSTATUS},1)
>
The referenced message is not related to the call at all. It occurs when an
"asterisk -r" console disconnects. As for hanging up a call when the remote
goes away that depends on the channel driver. For SIP both chan_sip and
chan_pjsip provide session timers which use SIP messages to determine if
the call is no longer valid, or RTP timeout which hangs up the call if
media is not flowing for a period of time. These are configured in the
respective channel driver configuration file.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?
Dovid Bender writes: > So long as the tcp socket is open your SBC should send the call back over > the same socket. Now it can be that your SBC is seeing the socket as > timing out. If you are using Kamailio you can have it send tcp keep alives > every so often so that the socket stays up. SBC? I am curious if the "reuse registration TCP connection" is required by standards or if it is merely obviously good practice. I have had this problem too with asterisk 16.5.0 This is not the first recommendation I have seen to use kamailio as a proxy for asterisk, for these sorts of issues as well as clients that change addresses. Unfortunately the "jsr pc, set_up_kamailo" subroutine call is still executing so I can't say if things work right then... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] SIP via TCP - new TCP session per call or use same session for multiple calls?
So long as the tcp socket is open your SBC should send the call back over the same socket. Now it can be that your SBC is seeing the socket as timing out. If you are using Kamailio you can have it send tcp keep alives every so often so that the socket stays up. On Fri, Dec 27, 2019 at 10:41 AM Benoit Panizzon wrote: > Hi List > > I wonder how SIP via TCP is supposed to work. Not realy Asterisk > related, but I hope you experts might be able to help out :-) > > One of our customers has a SIP device registering via a complex NAT. To > benefit from TCP Connection Tracking, he choose TCP instead of UDP. > > So he expected, that an incoming call would be sent back to him on the > already open TCP connection, making it easy to get through that NAT. > > This is not the case. Our SBC is attempting to initiate a new SIP TCP > connection towards the NAT Firewall of the customer thus getting > dropped because this is not the outgoing established connection opened > during the registration. > > So, how should SIP via TCP work? Should one TCP session be used for all > signaling of potentially multiple concurrent calls, as expected by our > customer. Or is it usual to make one TCP session per call as observed? > > Mit freundlichen Grüssen > > -Benoît Panizzon- > -- > I m p r o W a r e A G-Leiter Commerce Kunden > __ > > Zurlindenstrasse 29 Tel +41 61 826 93 00 > CH-4133 PrattelnFax +41 61 826 93 01 > Schweiz Web http://www.imp.ch > __ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- 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] Handling a non-responsive peer after it answers
I'm looking for a way of detecting in my dialplan when a peer becomes
non-responsive after answering. It seems that Asterisk knows when the peer
becomes non-responsive because it logs "Remote UNIX connection
disconnected" around the same time, and it seems that
if there is no following "Remote UNIX connection" within a short time, then
the peer can be considered non-responsive. Is there a way to configure a
handler for this state?
We use v14.7.6 and we dial the peer this way:
same =>
n,Set(CHANNEL(hangup_handler_push)=${CONTEXT},handleHangupByCaller,1(args))
same =>
n,Dial(${AddressToReachPeer},2,b(${CONTEXT}^afterDialingPeerLogIpOfCb^1(${UUID}^${StartEpoch})))
same => n,Goto(handle${DIALSTATUS},1)
Cheers,
David
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[asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?
Hi List I wonder how SIP via TCP is supposed to work. Not realy Asterisk related, but I hope you experts might be able to help out :-) One of our customers has a SIP device registering via a complex NAT. To benefit from TCP Connection Tracking, he choose TCP instead of UDP. So he expected, that an incoming call would be sent back to him on the already open TCP connection, making it easy to get through that NAT. This is not the case. Our SBC is attempting to initiate a new SIP TCP connection towards the NAT Firewall of the customer thus getting dropped because this is not the outgoing established connection opened during the registration. So, how should SIP via TCP work? Should one TCP session be used for all signaling of potentially multiple concurrent calls, as expected by our customer. Or is it usual to make one TCP session per call as observed? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
