Thanks all. I will certainly have a play with the dial plan and test.
I think that I initially removed some of my Answer lines when I was using the
Connected Caller ID as it didn't work in some instances.
Regards
Michael Knill
From: David Kerr <da...@kerr.net>
Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net>
Date: Tuesday, 14 March 2017 at 6:58 am
To: AstLinux List <astlinux-users@lists.sourceforge.net>
Subject: Re: [Astlinux-users] Astlinux on the edge
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<mailto: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<mailto: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.kn...@ipcsolutions.com.au<mailto:michael.kn...@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<mailto:li...@lonnie.abelbeck.com>>
> > Reply-To: AstLinux List
> > <astlinux-users@lists.sourceforge.net<mailto:astlinux-users@lists.sourceforge.net>>
> > Date: Tuesday, 14 March 2017 at 12:51 am
> > To: AstLinux List
> > <astlinux-users@lists.sourceforge.net<mailto: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<mailto: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<mailto:li...@lonnie.abelbeck.com>>
> >> Reply-To: AstLinux List
> >> <astlinux-users@lists.sourceforge.net<mailto:astlinux-users@lists.sourceforge.net>>
> >> Date: Thursday, 9 March 2017 at 1:22 am
> >> To: AstLinux List
> >> <astlinux-users@lists.sourceforge.net<mailto: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<http://192.168.0.0/255.255.0.0>'
> >> and "localnet=10.0.0.0/255.0.0.0<http://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<http://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<mailto: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<mailto:li...@lonnie.abelbeck.com>>
> >>> Reply-To: AstLinux List
> >>> <astlinux-users@lists.sourceforge.net<mailto:astlinux-users@lists.sourceforge.net>>
> >>> Date: Saturday, 4 March 2017 at 2:54 pm
> >>> To: AstLinux List
> >>> <astlinux-users@lists.sourceforge.net<mailto: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<mailto: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
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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<mailto:pay...@krisk.org>.
> >>>
> >>>
> >>> ------------------------------------------------------------------------------
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> >>> pay...@krisk.org<mailto: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<mailto: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
> >> _______________________________________________
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> >>
> >> Donations to support AstLinux are graciously accepted via PayPal to
> >> pay...@krisk.org<mailto: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
> > _______________________________________________
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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<mailto:pay...@krisk.org>.
> >
> >
> > ------------------------------------------------------------------------------
> > Check out the vibrant tech community on one of the world's most
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> >
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> > pay...@krisk.org<mailto:pay...@krisk.org>.
>
>
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>
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> http://sdm.link/slashdot_______________________________________________
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