I do exactly the same thing on inbound calls...
[ring-answer]
exten = s,1,ExecIf($["${CDR(answer)}"!=""]?Return())
same = n,Ringing()
same = n,Wait(2)
same = n,Answer()
same = n,Return()
In my case I chose two seconds and I put it into a context I
GoSub(ring-answer,s,1) too. I first test to make sure the line has not
already been answered (as I have nested IVRs so you may land somewhere in
an already answered state).
David
On Mon, Mar 13, 2017 at 3:34 PM, Lonnie Abelbeck <li...@lonnie.abelbeck.com>
wrote:
> Hi David,
>
> > Interesting that you Answer() before Dial() any local extensions. Is
> there a reason you do that?
>
> Obsessive consistency ? :-) I don't recall for certain, my home/office
> dialplan has not changed much for years so once it worked well I left it
> alone. But I have never had any issues with follow-me and such.
>
> Though, there are special cases where Answer() is not called such as
> blacklist "No Answer" but for blacklist "Zapateller" Answer() needs to be
> called.
>
> FYI, for inbound calls I usually start with an out-of-band Ring for 1+
> seconds, then answer and play 1 or 2 seconds of silence for everything to
> settle down.
> --
> [daytime-ivr]
> exten => s,1,Ringing
> exten => s,n,Wait(1.1)
> exten => s,n,Answer
> exten => s,n,Playback(silence/2)
> ... etc ...
> --
>
> > Requires https://issues.asterisk.org/jira/browse/ASTERISK-26587
>
> Very interesting David, thanks for sharing.
>
> Lonnie
>
>
>
> On Mar 13, 2017, at 1:42 PM, David Kerr <da...@kerr.net> wrote:
>
> > Lonnie,
> > Interesting that you Answer() before Dial() any local extensions. Is
> there a reason you do that? I don't.... I let the end extension do the
> answer. Now if there is no answer and I fall through to voicemail or if
> for some other reason connect into an IVR (I send all calls after midnight
> to an IVR) then I Answer() and start sending audio back to the caller.
> >
> > By-the-way... on voicemail. Remember the good old days of answering
> machines where you could listen to the caller record their message and
> decide wither to pick up while the person left a message. I have that
> working now with Asterisk... when a person leaves a voicemail message I can
> have a speakerphone (with autoanswer) act as a monitoring device so I can
> listen into the message. If I want to intercept, I can do so from any
> extension in my house. Really cool. Requires
> https://issues.asterisk.org/jira/browse/ASTERISK-26587 which is merged
> into Asterisk 15 and has patches for 13 and 11 attached to the issue. If
> anyone wants the dialplan magic for this let me know.
> >
> > David
> >
> > On Mon, Mar 13, 2017 at 1:01 PM, Lonnie Abelbeck <
> li...@lonnie.abelbeck.com> wrote:
> > Michael,
> >
> > Keeping Asterisk in the path is key, and calling Answer() is required at
> some point to do that.
> >
> > I always call Answer() before calling local phones, of course any IVR
> requires calling Answer() first.
> >
> > Though it may be possible, depending on your SIP trunk provider and
> enabling "directmedia=yes" for the trunk only, to selectively re-invice
> inbound calls back to the SIP trunk and not calling Answer(). Since this
> depends on your SIP trunk provider, it may work one day and stop working
> another day.
> >
> > If these kind of "hair-pin" calls are not common, play it safe and
> answer the call and dial back out, keeping Asterisk in the path.
> >
> > Lonnie
> >
> >
> > On Mar 13, 2017, at 11:33 AM, Michael Knill <michael.knill@ipcsolutions.
> com.au> wrote:
> >
> > > Yes thanks Lonnie
> > >
> > > No the call never gets to the IP Phone. I manage all my forwarding
> within the Asterisk dial plan. And yes Im always keeping Asterisk in the
> path but as prompted by David, I suspect now that Asterisk is not bridging
> the call as I never actually Answer it in my dial plan.
> > >
> > > We will see.
> > >
> > > Regards
> > > Michael Knill
> > >
> > > -----Original Message-----
> > > From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
> > > Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net>
> > > Date: Tuesday, 14 March 2017 at 12:51 am
> > > To: AstLinux List <astlinux-users@lists.sourceforge.net>
> > > Subject: Re: [Astlinux-users] Astlinux on the edge
> > >
> > > Michael,
> > >
> > > I hope others here will offer their SIP experiences, but can you
> define in more detail what the failure mode is. I'll guess a little ...
> > >
> > > A call comes in via your SIP trunk provider, dials a local extension,
> either the extensions is busy (or DND set) or no answer then the Asterisk
> dialplan does what ?
> > >
> > > Or are you using a "feature" of the IP Phone to initiate the outbound
> call when DND or other is set ? Using Asterisk as the server or directly
> to the SIP trunk provider ?
> > >
> > > Explain exactly who does what and when.
> > >
> > > Bottom line, when behind NAT keep Asterisk in the path at all times.
> Possibly in your failure case your IP Phone is re-inviting around Asterisk ?
> > >
> > > Lonnie
> > >
> > >
> > > On Mar 13, 2017, at 4:32 AM, Michael Knill <
> michael.kn...@ipcsolutions.com.au> wrote:
> > >
> > >> Ok my initial NAT testing is exhibiting the following issue which I
> remember previously occurred.
> > >> Calls to and from extensions to external are fine with the below
> configuration.
> > >> The failure scenario however is an incoming call forwarding out to an
> external call (hair pin) where there is no audio both ways.
> > >>
> > >> I spend ages trying to troubleshoot the issue to no avail. I looked
> though all the SIP SDP trying to work out what is happening. What I don't
> quite understand, and I am hoping all the SIP experts can help, is that I
> don't have any ALG’s set up so how does the external proxy know what media
> port to connect to? I understand that rport is sent in the Via header which
> gives the external address but this seems like its only for signalling!
> > >>
> > >> What is interesting is that I do a packet sniff on the router
> external interface (Mikrotik) and I don't see ANY RTP packets hitting or
> exiting. What is also interesting is that when I answer the incoming call
> from an extension and transfer it externally, the media works fine.
> > >> I suspect it has something to do with this which I cant seem to find
> any info on:
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- SIP/gwy2-00000037 is making progress passing it to
> Local/0400113919@DialPlan1-00000025;2
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >> -- Local/0400113919@DialPlan1-00000025;2 requested media update
> control 20, passing it to SIP/gwy2-00000037
> > >>
> > >> Any ideas? No NAT for me currently until I can fix this.
> > >>
> > >> Regards
> > >> Michael Knill
> > >>
> > >> -----Original Message-----
> > >> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
> > >> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net>
> > >> Date: Thursday, 9 March 2017 at 1:22 am
> > >> To: AstLinux List <astlinux-users@lists.sourceforge.net>
> > >> Subject: Re: [Astlinux-users] Astlinux on the edge
> > >>
> > >> Michael,
> > >>
> > >> If you place AstLinux behind a NAT firewall as a PBX ...
> > >>
> > >> -- No NAT port forwarding to AstLinux (except for possible OpenVPN
> for remote IP Phones) and disable any upstream SIP ALG's.
> > >>
> > >> -- Set "directmedia=no" for all phones and the trunk, all media goes
> through Asterisk
> > >>
> > >> -- Set "qualify=yes" on trunk SIP peer to keep the upstream firewall
> state active
> > >>
> > >> -- Set "nat=force_rport,comedia" on the trunk SIP peer to force NAT
> handling, the only peer that does NAT to Asterisk
> > >>
> > >> -- Set "localnet=192.168.0.0/255.255.0.0' and "localnet=
> 10.0.0.0/255.0.0.0" to cover any LAN and OpenVPN networks which are not
> NAT'ed to Asterisk.
> > >>
> > >> -- When using remote IP Phones over OpenVPN, since asterisk will bind
> to the openvpn server tun interface, use the openvpn network (possibly
> 10.8.0.0/24) for tunneled SIP endpoints.
> > >>
> > >> (Readers, if I have missed or mangled any of the above, please
> correct.)
> > >>
> > >> Bottom line, an AstLinux PBX behind NAT should be workable for
> production.
> > >>
> > >> Lonnie
> > >>
> > >>
> > >> On Mar 7, 2017, at 8:01 PM, Michael Knill <
> michael.kn...@ipcsolutions.com.au> wrote:
> > >>
> > >>> Hi thanks Lonnie. Sorry this went into my junk for some reason.
> > >>>
> > >>> 1) Yes this is certainly a problem but I have also experienced
> problems with no media on calls being hairpinned through Asterisk from the
> external trunk. This may be solvable with port forwarding however. Maybe I
> should do some testing on this and specify some known and working
> router/firewall configurations.
> > >>> 2) I use Open VPN for my external phones so it could be solved this
> way.
> > >>>
> > >>> I am currently negotiating with the partner and it looks like they
> will take option 3 below which I think is the best compromise.
> > >>>
> > >>> Regards
> > >>> Michael Knill
> > >>>
> > >>> -----Original Message-----
> > >>> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
> > >>> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net>
> > >>> Date: Saturday, 4 March 2017 at 2:54 pm
> > >>> To: AstLinux List <astlinux-users@lists.sourceforge.net>
> > >>> Subject: Re: [Astlinux-users] Astlinux on the edge
> > >>>
> > >>> Hi Michael,
> > >>>
> > >>> My guess is "it depends" ... your IT partners go into a auto repair
> shop with a 5 year old residential-grade router, etc. (ie. a mess) then
> making AstLinux the edge device would be a major upgrade, not to mention
> the added voice functionality.
> > >>>
> > >>> Then again your IT partners go into a dentist's office which were
> previously sold more router than they needed, it may not seem right to put
> AstLinux in front of it.
> > >>>
> > >>> My guess is you need to plan for both situations.
> > >>>
> > >>> A couple comments ...
> > >>>
> > >>> 1) If AstLinux will only serve SIP endpoints on the private side, no
> roaming public endpoints, then being behind NAT is workable, only the trunk
> is effected by NAT. Always disable any upstream SIP ALG's, almost always
> bad news. Keep in mind no upstream port-forwarding is needed for this
> scenario, and always keep the AstLinux firewall enabled for the Adaptive
> Ban and other protections to be kept in place.
> > >>>
> > >>> 2) Else if roaming public endpoints need to be supported, placing
> AstLinux at the edge will make SIP easier. AstLinux comes with a dmz-dnat
> plugin, the idea is to move a pre-existing router from the WAN to
> AstLinux's LAN with a static IP address and configure the plugin which
> internally performs a " -j DNAT --to-destination $DMZ_IP " *all* traffic
> not allowed directly into AstLinux. WARNING - this plugin was written many
> years ago and has not been tested as thoroughly as I would like to see for
> production purposes. Though if there are issues with the dmz-dnat plugin
> they could be remedied.
> > >>>
> > >>> Lonnie
> > >>>
> > >>>
> > >>> On Mar 3, 2017, at 4:50 PM, Michael Knill <
> michael.kn...@ipcsolutions.com.au> wrote:
> > >>>
> > >>>> Hi all
> > >>>>
> > >>>> Im looking to push my Astlinux business this year and this will
> rely heavily on partners. These partners will usually be IT Service
> providers that have a number of small business customers and that they want
> to add voice as a value add product.
> > >>>>
> > >>>> Now here is where the problem lies. Most of these providers would
> currently be maintaining the site firewall but as Astlinux is designed to
> be on the edge, its an issue. So what do you do?
> > >>>> 1) Put Astlinux in front of their firewall and open up the
> necessary ports and protocols. The problem here is that they lose
> flexibility in what they can do and there is another provider in the mix.
> Its also a problem if they are retailing the broadband connection for the
> site with too many dependencies.
> > >>>> 2) Put their firewall on an Astlinux DMZ with a public IP
> Address. They now have more flexibility and I can control Qos. Still issues
> with being reliant on another provider and additional IP Addresses can be
> expensive or unobtainable. I assume I can actually do this with Astlinux!
> > >>>> 3) Put Astlinux as a DMZ in their firewall with a public IP
> Address. They now have complete control however QoS would need to be
> configured on the firewall and additional IP Addresses can be expensive or
> unobtainable. PS this is the model I have with one of my partners
> > >>>> 4) Sit behind the firewall and rely on port forwarding and/or
> ALG’s. Inviting trouble but possible if you have a known working
> configuration
> > >>>>
> > >>>> Im interested to know what others are doing in this space.
> > >>>>
> > >>>> Regards
> > >>>> Michael Knill
> > >>>
> > >>>
> > >>>
> > >>>
> > >>> ------------------------------------------------------------
> ------------------
> > >>> Check out the vibrant tech community on one of the world's most
> > >>> engaging tech sites, SlashDot.org! http://sdm.link/slashdot
> > >>> _______________________________________________
> > >>> Astlinux-users mailing list
> > >>> Astlinux-users@lists.sourceforge.net
> > >>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> > >>>
> > >>> Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
> > >>>
> > >>>
> > >>> ------------------------------------------------------------
> ------------------
> > >>> Announcing the Oxford Dictionaries API! The API offers world-renowned
> > >>> dictionary content that is easy and intuitive to access. Sign up for
> an
> > >>> account today to start using our lexical data to power your apps and
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> > >>>
> > >>> Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
> > >>
> > >>
> > >> ------------------------------------------------------------
> ------------------
> > >> Announcing the Oxford Dictionaries API! The API offers world-renowned
> > >> dictionary content that is easy and intuitive to access. Sign up for
> an
> > >> account today to start using our lexical data to power your apps and
> > >> projects. Get started today and enter our developer competition.
> > >> http://sdm.link/oxford
> > >> _______________________________________________
> > >> Astlinux-users mailing list
> > >> Astlinux-users@lists.sourceforge.net
> > >> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> > >>
> > >> Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
> > >>
> > >>
> > >> ------------------------------------------------------------
> ------------------
> > >> Announcing the Oxford Dictionaries API! The API offers world-renowned
> > >> dictionary content that is easy and intuitive to access. Sign up for
> an
> > >> account today to start using our lexical data to power your apps and
> > >> projects. Get started today and enter our developer competition.
> > >> http://sdm.link/oxford
> > >> _______________________________________________
> > >> Astlinux-users mailing list
> > >> Astlinux-users@lists.sourceforge.net
> > >> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> > >>
> > >> Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
> > >
> > >
> > > ------------------------------------------------------------
> ------------------
> > > Announcing the Oxford Dictionaries API! The API offers world-renowned
> > > dictionary content that is easy and intuitive to access. Sign up for an
> > > account today to start using our lexical data to power your apps and
> > > projects. Get started today and enter our developer competition.
> > > http://sdm.link/oxford
> > > _______________________________________________
> > > Astlinux-users mailing list
> > > Astlinux-users@lists.sourceforge.net
> > > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> > >
> > > Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
> > >
> > >
> > > ------------------------------------------------------------
> ------------------
> > > Check out the vibrant tech community on one of the world's most
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> > > _______________________________________________
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> > > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> > >
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> pay...@krisk.org.
> >
> >
> > ------------------------------------------------------------
> ------------------
> > Check out the vibrant tech community on one of the world's most
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> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
> >
> > ------------------------------------------------------------
> ------------------
> > Check out the vibrant tech community on one of the world's most
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> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
>
>
> ------------------------------------------------------------
> ------------------
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>
> Donations to support AstLinux are graciously accepted via PayPal to
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