Re: [asterisk-users] Get context with hangup handler
It seems like this would work and I appreciate the code but I doubt it
would make it into the main branch. I think I am stuck with setting it in
the context.
On Thu, Jan 6, 2022 at 8:26 AM wrote:
> I'm not sure what other implications this might have, but does something
> like this work for you? You would need to apply the following patch[1].
>
> Manually trying to save the last context/exten/etc. in the dialplan itself
> is guaranteed to be an ugly solution. Let Asterisk do it for you.
>
> [f1]
> exten => s,1,NoOp(${CONTEXT})
> same => n,NoOp(${CONTEXT} / ${LASTCONTEXT})
> same => n,Goto(f2,s,1)
> [f2]
> exten => s,1,NoOp(${CONTEXT} / ${LASTCONTEXT})
> same => n,NoOp(${CONTEXT} / ${LASTCONTEXT})
> same => n,Hangup()
>
> pbxdev*CLI> channel originate Local/s@f1 application Wait 30
> [Jan 6 08:19:30] -- Called s@f1
> [Jan 6 08:19:30] -- Executing [s@f1:1] NoOp("Local/s@f1-0002;2",
> "f1") in new stack
> [Jan 6 08:19:30] -- Executing [s@f1:2] NoOp("Local/s@f1-0002;2",
> "f1 / ") in new stack
> [Jan 6 08:19:30] -- Executing [s@f1:3] Goto("Local/s@f1-0002;2",
> "f2,s,1") in new stack
> [Jan 6 08:19:30] -- Goto (f2,s,1)
> [Jan 6 08:19:30] -- Executing [s@f2:1] NoOp("Local/s@f1-0002;2",
> "f2 / f1") in new stack
> [Jan 6 08:19:30] -- Executing [s@f2:2] NoOp("Local/s@f1-0002;2",
> "f2 / f1") in new stack
> [Jan 6 08:19:30] -- Executing [s@f2:3] Hangup("Local/s@f1-0002;2",
> "") in new stack
> [Jan 6 08:19:30] == Spawn extension (f2, s, 3) exited non-zero on
> 'Local/s@f1-0002;2'
>
> [1] https://code.phreaknet.org/asterisk/lastcontext.diff
> On 1/5/2022 10:22 PM, Dovid Bender wrote:
>
> Steve,
>
> I thought of this but that would mean I would need to add this to the
> beginning of every context which I can do, but I was trying to avoid.
>
>
> On Wed, Jan 5, 2022 at 10:06 PM Steve Edwards
> wrote:
>
>> On Wed, 5 Jan 2022, Steve Edwards wrote:
>>
>> > same = n, set(LAST-CONTEXT=${context}
>>
>> Double damn. I munged the case on ${CONTEXT}. I give up for today :)
>>
>> --
>> Thanks in advance,
>> -
>> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
>> https://www.linkedin.com/in/steve-edwards-4244281
>>
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>> _
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>
>
>
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Re: [asterisk-users] asterisk and maybe a freepbx question
OK, that tells me something, I will disable pjsit for now, learn about it and try again. On Sun, 09 Jan 2022 06:39:55 -0500, John Harragin wrote: > > [1 ] > [1.1 ] > You can also set up multiple physical or vlan(ed) interfaces and bind sip > to one and pjsip to the other - then you have to set up the appropriate > interface routing too for both inbound and outbound packets which takes a > good understanding of your network topology and the locations of your > respective devices. You might be able to do it with multiple addresses on > your interface too (although I haven't tried it). > > All of the packets have to be presented to the appropriate channel > otherwise get discarded. You can't set it up so if a packet is from a > device not registered with pjsip, it gets passed to chan_sip to try. > > For me, I had both channel types running on production machines while I > migrated to pjsip or when not being able to figure out how to set up some > property in pjsip that you had running in sip. Each time I've had to do > this, eventually I was able get it all running within pjsip. I also already > had multiple vlans configured for my servers (with voip exclusive to one). > > The short story is that it is easier to learn how to get things working > within pjsip than learning the tricky networking setup. > > > On Sun, Jan 9, 2022 at 2:49 AM Duncan Turnbull > wrote: > > > > > > > > > > > > On 9/01/2022, at 7:11 PM, John Covici wrote: > > > > > > On Sat, 08 Jan 2022 19:17:57 -0500, > > > Antony Stone wrote: > > >> > > >>> On Sunday 09 January 2022 at 00:50:27, John Covici wrote: > > >>> > > >>> Hi. I am using asterisk 18.3 and freepbx. > > >> > > >> Hm, which version of FreePBX uses Asterisk 18.3? > > >> > > >>> How can both sip and pjsip be listening at port 5060 at the same time > > >> > > >> They can't. > > >> > > >> One might be on TCP and the other on UDP, but you can't have them both > > >> listening on the same port with the same protocol. > > > > In freepbx you enable chan sip or pjsip or both and set what ports they use > > > > The choices are either in advanced settings or sip settings > > > > Disable pjsip and reset the chan_sip port to 5060 or use pjsip. With them > > both enabled sometimes odd things happen but it will still work. You will > > get lots of error messages though > > > > > > >> > > >>> for instance I get: > > >>> > > >>> [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: > > >>> > > SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" > > >>> > > Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 > > >>> 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060 > > ",RemoteAddress="IPV4/UDP/ > > >>> 45.134.144.118/5823 > > ",ACLName="registrar_attempt_without_configured_aors" > > >> > > >> What makes you think chan_sip and pjsip are both listening on UDP 5060? > > >> > > >>> I would like pjsit not to listen,till I figure out how to configure > > >>> the thing, so my logs don't fill up with messages. > > >>> > > >>> Thanks in advance for any suggestions. > > >> > > >> As far as I recall using FreePBX, there is a selector for the SIP > > protocol to > > >> tell it whether you want it to use pjsip or chan_sip. I don't think it > > even > > >> supports using both at the same time, so simply make sure that is set > > to > > >> chan_sip and you should be fine. > > >> > > >> On the other hand, why do you need to learn "how to configure the > > thing" if > > >> you're using FreePBX? Part of the whole point is that it does the > > fiddly > > >> techie sutff in the background for you, and you just need to use the > > personnel- > > >> department-friendly web GUI. > > > > > > This is what I thought as well, I just generated one trunk using the > > > old chan_sip and expected nothing from pjsit, yet I get all kinds of > > > errors like > > > [2022-01-08 17:08:59] WARNING[487628] res_pjsip_registrar.c: Endpoint > > > 'anonymous' (45.134.144.118:5823) has no configured AORs > > > > > > so I am very confused as to why this is happening. > > > > > > -- > > > Your life is like a penny. You're going to lose it. The question is: > > > How do > > > you spend it? > > > > > > John Covici wb2una > > > cov...@ccs.covici.com > > > > > > -- > > > _ > > > -- 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
