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.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> > > 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 > >>> 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. > > > > > > ------------------------------------------------------------------------------ > > 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 > > 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. > > > ------------------------------------------------------------------------------ > 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. > > ------------------------------------------------------------------------------ > 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. ------------------------------------------------------------------------------ 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.
