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.knill@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.knill@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.knill@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
> >>>
> >>>
> >>>
> >>>
> >>> ------------------------------------------------------------
> ------------------
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> >>
> >>
> >> ------------------------------------------------------------
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> >> http://sdm.link/oxford
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> >> Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
> >>
> >>
> >> ------------------------------------------------------------
> ------------------
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> >> 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
> >> _______________________________________________
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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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> > 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
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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
> pay...@krisk.org.
>
>
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>
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>
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