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Re: Amportal-users digest, Vol 1 #473 - 12 msgs: msg#00206

Subject: Re: Amportal-users digest, Vol 1 #473 - 12 msgs
Hi,

 Anybody has any experience on Video support--the type of SIP video phone and 
configuration?

Thanks in advance!

Nick

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> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Amportal-users digest..."
> 
> 
> Today's Topics:
> 
>    1. Capi? (Mona Meyer)
>    2. soumission (gervais janvier)
>    3. can I get documentation on using amp 1.09 beta1? (hank smith)
>    4. After 1.10.007->1.10.009.1 upgrade internal calls are no longer
>        working? (Remco Barende)
>    5. Which config files are automatically written? (Michael Vogel)
>    6. Re: Which config files are automatically written? (Ariel Batista)
>    7. Re: Which config files are automatically written? (Michael Vogel)
>    8. Re: Which config files are automatically written? (Ariel Batista)
>    9. Re: Which config files are automatically written? ([ZaglerNetWorks])
>   10. Re: Which config files are automatically written? (Michael Vogel)
>   11. Re: Which config files are automatically written? (Michael Vogel)
> 
> --__--__--
> 
> Message: 1
> Date: Mon, 29 Aug 2005 08:24:53 +0200
> From: Mona Meyer <meymona-Re5JQEeQqe8AvxtiuMwx3w@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: [Amportal-users] Capi?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> Hi,
> 
> I managed to install AMP on Sarge and I am really impressed about it.
> But I did not find a "how to implement capi or HFC" yet. Is there one?
> 
> 
> --__--__--
> 
> Message: 2
> Date: Mon, 29 Aug 2005 09:29:21 +0200 (CEST)
> From: gervais janvier <xyzjanvier-Qt13gs6zZMY@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: [Amportal-users] soumission
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> --0-1208693890-1125300561=:12995
> Content-Type: text/plain; charset=iso-8859-1
> Content-Transfer-Encoding: 8bit
> 
> soumission
> 
>               
> ---------------------------------
>  Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
>  T?l?chargez le ici !  
> --0-1208693890-1125300561=:12995
> Content-Type: text/html; charset=iso-8859-1
> Content-Transfer-Encoding: 8bit
> 
> <DIV>soumission</DIV><p>
>               <hr size=1> 
> <b><font color=#FF0000>Appel audio GRATUIT</font> partout dans le monde</b> 
> avec le nouveau Yahoo! Messenger<br> 
> <a 
> href="http://us.rd.yahoo.com/messenger/mail_taglines/default/*http://fr.messenger.yahoo.com";>T?l?chargez
>  le ici !</a> 
>  
> 
> --0-1208693890-1125300561=:12995--
> 
> 
> --__--__--
> 
> Message: 3
> From: "hank smith" <hanksmith4-ihVZJaRskl1bRRN4PJnoQQ@xxxxxxxxxxxxxxxx>
> To: <amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx>
> Date: Mon, 29 Aug 2005 01:25:06 -0700
> Subject: [Amportal-users] can I get documentation on using amp 1.09 beta1?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> hello can I please get some documentation on configuring amp 1.09 beta1?
> I am unsure how to configure my phone with the new oppsions.
> thanks
> hank
> 
> 
> --__--__--
> 
> Message: 4
> Date: Mon, 29 Aug 2005 13:54:33 +0200 (CEST)
> From: Remco Barende <asterisk-zEI9Lo81CwePKud3HExfWg@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: [Amportal-users] After 1.10.007->1.10.009.1 upgrade internal calls 
> are no longer
>  working?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> Hi list
> 
> I upgraded AMP from 1.10.0007 to 10.10.009.1 but now internal calls 
> stopped working. If you call an internal extension you either get a busy 
> signal or nothing, in the case of nothing the called party does see the 
> number in the display (no name) but the phone isn't ringing.
> 
> Some snippets from the * console which I think are relevant:
> 
> Use of uninitialized value in length at 
> /var/lib/asterisk/agi-bin/dialparties.agi line 113, <STDIN> line 41.
>         >  dialparties.agi: extnum: 232
>         >  dialparties.agi: exthascw: 0
>         >  dialparties.agi: exthascfb: 0
> Use of uninitialized value in concatenation (.) or string at 
> /var/lib/asterisk/agi-bin/dialparties.agi line 119, <STDIN> line 44.
>         >  dialparties.agi: extcfb:
>      --  dialparties.agi: Checking CW and CFB status for extension 232
>    == Parsing '/etc/asterisk/manager.conf': Found
>    == Parsing '/etc/asterisk/manager_custom.conf': Not found (No such file 
> or directory)
>    == Manager 'admin' logged on from 127.0.0.1
>    == Manager 'admin' logged off from 127.0.0.1
>         >  dialparties.agi: extstate: -1
>      --  dialparties.agi: ExtensionState for 232 could not be 
> read...assuming ok
> Argument "" isn't numeric in numeric ne (!=) at 
> /var/lib/asterisk/agi-bin/dialparties.agi line 151, <STDIN> line 48.
> Use of uninitialized value in concatenation (.) or string at 
> /var/lib/asterisk/agi-bin/dialparties.agi line 234, <STDIN> line 50.
>      --  dialparties.agi: DbSet CALLTRACE/232 to 228
>      -- AGI Script Executing Application: (NoOp) Options: ()
>      -- AGI Script dialparties.agi completed, returning 0
>      -- Executing NoOp("SIP/228-fe9c", "Returned from dialparties with no 
> extensions to call") in new stack
>      -- Executing SetVar("SIP/228-fe9c", "DIALSTATUS=BUSY") in new stack
>      -- Executing GotoIf("SIP/228-fe9c", "0?s-BUSY|1") in new stack
>      -- Executing GotoIf("SIP/228-fe9c", "1?s-BUSY|1") in new stack
>      -- Goto (macro-exten-vm,s-BUSY,1)
>      -- Executing NoOp("SIP/228-fe9c", "Extension is reporting BUSY and has 
> no Voicemail") in new stack
>      -- Executing Busy("SIP/228-fe9c", "") in new stack
>    == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 
> 'SIP/228-fe9c' in macro 'exten-vm'
>    == Spawn extension (from-internal, 232, 1) exited non-zero on 
> 'SIP/228-fe9c'
>      -- Executing Macro("SIP/228-fe9c", "hangupcall") in new stack
>      -- Executing ResetCDR("SIP/228-fe9c", "w") in new stack
>      -- Executing NoCDR("SIP/228-fe9c", "") in new stack
>      -- Executing Wait("SIP/228-fe9c", "5") in new stack
>    == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 
> 'SIP/228-fe9c' in macro 'hangupcall'
>    == Spawn extension (from-internal, h, 1) exited non-zero on 
> 'SIP/228-fe9c'
> 
> 
> 
> 
> --__--__--
> 
> Message: 5
> Date: Mon, 29 Aug 2005 16:54:00 +0200
> From: Michael Vogel <icarus-Rk72x6omD0g@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: [Amportal-users] Which config files are automatically written?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> Hi!
> 
> I'm playing with Asterisk@Home by now. Since I need some modification I 
> have to manually change some config files.
> 
> Which config files can I change manually? Which config files are written 
> by the system?
> 
> Bye!
> 
> Michael
> 
> 
> --__--__--
> 
> Message: 6
> Date: Mon, 29 Aug 2005 11:16:34 -0400
> From: Ariel Batista <arielb27-Re5JQEeQqe8AvxtiuMwx3w@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: Re: [Amportal-users] Which config files are automatically written?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> The files you can change are all the customs.conf files. If you make
> changes to any of the additional.conf they will be replaced by amp.=20
> But as all things are the main files like extensions.conf sip.conf and
> iax2.conf don't get changed unless you upgrade the main system. Which
> you can also do some minor changes to them if needed.  But keep this
> changes handy after you upgrade so you can add your changes to the
> files.
> 
> Ariel
> 
> On 8/29/05, Michael Vogel <icarus-Rk72x6omD0g@xxxxxxxxxxxxxxxx> wrote:
> > Hi!
> >=20
> > I'm playing with Asterisk@Home by now. Since I need some modification I
> > have to manually change some config files.
> >=20
> > Which config files can I change manually? Which config files are written
> > by the system?
> >=20
> > Bye!
> >=20
> > Michael
> >=20
> >=20
> > -------------------------------------------------------
> > SF.Net email is Sponsored by the Better Software Conference & EXPO
> > September 19-22, 2005 * San Francisco, CA * Development Lifecycle Practic=
> es
> > Agile & Plan-Driven Development * Managing Projects & Teams * Testing & Q=
> A
> > Security * Process Improvement & Measurement * http://www.sqe.com/bsce5sf
> > _______________________________________________
> > Amportal-users mailing list
> > Amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> > https://lists.sourceforge.net/lists/listinfo/amportal-users
> >=20
> 
> 
> --=20
> Ariel Batista
> Kasi International
> Ph: 305-574-1495  FX: 305-574-0212
> Iaxtel: 17003838750 FWD: 65342
> 
> 
> --__--__--
> 
> Message: 7
> Date: Mon, 29 Aug 2005 17:44:20 +0200
> From: Michael Vogel <icarus-Rk72x6omD0g@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: Re: [Amportal-users] Which config files are automatically written?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> Ariel Batista schrieb:
> > The files you can change are all the customs.conf files.
> 
> It seems to be that my wishes can't be covered by just editing the=20
> customs-config files. My wishes are:
> 
> - Instead of the voicebox the caller should decide to get the voicebox=20
> or to get transfered to some central number (so he can decide if he=20
> wants to speak with a machine or a person).
> 
> - If somebody isn't reachable and hasn't enabled his voicebox the call=20
> should be transfered to the central number.
> 
> - Callers from inside should neither be terminated onto a voicebox nor=20
> be transfered to the central number. They should get the busy tone if=20
> somebody isn't reachable.
> 
> - Using two HFC-cards on an "Anlagenanschlu=DF" (I don't know the=20
> technical translation for this)
> 
> > If you make
> > changes to any of the additional.conf they will be replaced by amp.=20
> 
> Okay. So anything with "additional" in its name is written by AMP.
> 
> > But as all things are the main files like extensions.conf sip.conf and
> > iax2.conf don't get changed unless you upgrade the main system. Which
> > you can also do some minor changes to them if needed. But keep this
> > changes handy after you upgrade so you can add your changes to the
> > files.
> 
> I have to see how to make my changes as minor as possible. First of all=20
> I need a lot more knowledge about scripting ...
> 
> Bye!
> 
> Michael
> 
> 
> --__--__--
> 
> Message: 8
> Date: Mon, 29 Aug 2005 11:52:30 -0400
> From: Ariel Batista <arielb27-Re5JQEeQqe8AvxtiuMwx3w@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: Re: [Amportal-users] Which config files are automatically written?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> It's funny but most of what you need can be done with ring groups and
> different did routing. You can also create your own custom menu via
> amp. If you want a caller to get access to a dial tone just make a
> menu items that sends them to a custom menu that adds disa and
> requires a password.  You can send calls via the ring groups to local
> number like another phone number 3055551212# this will send it to your
> outbound dialing rules.
> 
> There is allot you can do with amp.  List the flow you want and let
> see if we can assist you with it's setup.
> 
> Ariel
> 
> On 8/29/05, Michael Vogel <icarus-Rk72x6omD0g@xxxxxxxxxxxxxxxx> wrote:
> > Ariel Batista schrieb:
> > > The files you can change are all the customs.conf files.
> >=20
> > It seems to be that my wishes can't be covered by just editing the
> > customs-config files. My wishes are:
> >=20
> > - Instead of the voicebox the caller should decide to get the voicebox
> > or to get transfered to some central number (so he can decide if he
> > wants to speak with a machine or a person).
> >=20
> > - If somebody isn't reachable and hasn't enabled his voicebox the call
> > should be transfered to the central number.
> >=20
> > - Callers from inside should neither be terminated onto a voicebox nor
> > be transfered to the central number. They should get the busy tone if
> > somebody isn't reachable.
> >=20
> > - Using two HFC-cards on an "Anlagenanschlu=DF" (I don't know the
> > technical translation for this)
> >=20
> > > If you make
> > > changes to any of the additional.conf they will be replaced by amp.
> >=20
> > Okay. So anything with "additional" in its name is written by AMP.
> >=20
> > > But as all things are the main files like extensions.conf sip.conf and
> > > iax2.conf don't get changed unless you upgrade the main system. Which
> > > you can also do some minor changes to them if needed. But keep this
> > > changes handy after you upgrade so you can add your changes to the
> > > files.
> >=20
> > I have to see how to make my changes as minor as possible. First of all
> > I need a lot more knowledge about scripting ...
> >=20
> > Bye!
> >=20
> > Michael
> >=20
> >=20
> > -------------------------------------------------------
> > SF.Net email is Sponsored by the Better Software Conference & EXPO
> > September 19-22, 2005 * San Francisco, CA * Development Lifecycle Practic=
> es
> > Agile & Plan-Driven Development * Managing Projects & Teams * Testing & Q=
> A
> > Security * Process Improvement & Measurement * http://www.sqe.com/bsce5sf
> > _______________________________________________
> > Amportal-users mailing list
> > Amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> > https://lists.sourceforge.net/lists/listinfo/amportal-users
> >=20
> 
> 
> --=20
> Ariel Batista
> Kasi International
> Ph: 305-574-1495  FX: 305-574-0212
> Iaxtel: 17003838750 FWD: 65342
> 
> 
> --__--__--
> 
> Message: 9
> Date: Mon, 29 Aug 2005 18:03:21 +0200
> From: "[ZaglerNetWorks]" <info-SHM7L3vmMwusTnJN9+BGXg@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: Re: [Amportal-users] Which config files are automatically written?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> for hfc-cards use bristuff from http://www.junghanns.net
> "anlagenanschluss" and "mehrger?teanschluss" works both
> 
> regards,
> alex
> 
> Michael Vogel schrieb:
> 
> > Ariel Batista schrieb:
> >
> >> The files you can change are all the customs.conf files.
> >
> >
> > It seems to be that my wishes can't be covered by just editing the 
> > customs-config files. My wishes are:
> >
> > - Instead of the voicebox the caller should decide to get the voicebox 
> > or to get transfered to some central number (so he can decide if he 
> > wants to speak with a machine or a person).
> >
> > - If somebody isn't reachable and hasn't enabled his voicebox the call 
> > should be transfered to the central number.
> >
> > - Callers from inside should neither be terminated onto a voicebox nor 
> > be transfered to the central number. They should get the busy tone if 
> > somebody isn't reachable.
> >
> > - Using two HFC-cards on an "Anlagenanschlu? (I don't know the 
> > technical translation for this)
> >
> >> If you make
> >> changes to any of the additional.conf they will be replaced by amp. 
> >
> >
> > Okay. So anything with "additional" in its name is written by AMP.
> >
> >> But as all things are the main files like extensions.conf sip.conf and
> >> iax2.conf don't get changed unless you upgrade the main system. Which
> >> you can also do some minor changes to them if needed. But keep this
> >> changes handy after you upgrade so you can add your changes to the
> >> files.
> >
> >
> > I have to see how to make my changes as minor as possible. First of 
> > all I need a lot more knowledge about scripting ...
> >
> > Bye!
> >
> > Michael
> >
> >
> > -------------------------------------------------------
> > SF.Net email is Sponsored by the Better Software Conference & EXPO
> > September 19-22, 2005 * San Francisco, CA * Development Lifecycle 
> > Practices
> > Agile & Plan-Driven Development * Managing Projects & Teams * Testing 
> > & QA
> > Security * Process Improvement & Measurement * http://www.sqe.com/bsce5sf
> > _______________________________________________
> > Amportal-users mailing list
> > Amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> > https://lists.sourceforge.net/lists/listinfo/amportal-users
> >
> 
> 
> 
> --__--__--
> 
> Message: 10
> Date: Mon, 29 Aug 2005 18:16:25 +0200
> From: Michael Vogel <icarus-Rk72x6omD0g@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: Re: [Amportal-users] Which config files are automatically written?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> Hi!
> 
> Ariel Batista schrieb:
> > It's funny but most of what you need can be done with ring groups and
> > different did routing. 
> 
> Oh it can? Great. I would like to change the system as less as possible 
> - of course.
> 
> > There is allot you can do with amp. List the flow you want and let
> > see if we can assist you with it's setup.
> 
> Okay, here it goes:
> 
> - All calls are coming in over ISDN (there will be no SIP or IAX-Outbound)
> - The calls should go to the different extensions (no central pickup, no 
> central voice menu or such things) (That means calls to 0815/4711-10 go 
> to extension 10, calls to 0815/4711-20 are going to 20, ...)
> - Everybody in the room should be able to pickup the call (I can make 
> this with ringgroups I think)
> - If the call isn't answered for some seconds the call should be 
> transferred to some central number (The central phone should be able to 
> do a pickup with every incoming call)
> - If the User has enabled the voicebox the caller should get a menu 
> where he can choose to get to the central number or to speak a message 
> on the voicebox
> 
> If you can help me by setting this up I would be very happy! (Since I 
> said my boss that A@H (which includes AMP) would be better than any 
> other solution that can be bought.)
> 
> Thx!
> 
> Michael
> 
> 
> --__--__--
> 
> Message: 11
> Date: Mon, 29 Aug 2005 19:54:01 +0200
> From: Michael Vogel <icarus-Rk72x6omD0g@xxxxxxxxxxxxxxxx>
> To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> Subject: Re: [Amportal-users] Which config files are automatically written?
> Reply-To: amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> 
> Michael Vogel schrieb:
> > 
> > - All calls are coming in over ISDN (there will be no SIP or IAX-Outbound)
> > - The calls should go to the different extensions (no central pickup, no 
> > central voice menu or such things) (That means calls to 0815/4711-10 go 
> > to extension 10, calls to 0815/4711-20 are going to 20, ...)
> > - Everybody in the room should be able to pickup the call (I can make 
> > this with ringgroups I think)
> > - If the call isn't answered for some seconds the call should be 
> > transferred to some central number (The central phone should be able to 
> > do a pickup with every incoming call)
> > - If the User has enabled the voicebox the caller should get a menu 
> > where he can choose to get to the central number or to speak a message 
> > on the voicebox
> 
> I forgot something: All this rules are only valid for external callers. 
> All calls from the inside should just terminate if the called person 
> doesn't answer or is busy.
> 
> Bye!
> 
> Michael
> 
> 
> 
> --__--__--
> 
> _______________________________________________
> Amportal-users mailing list
> Amportal-users-5NWGOfrQmneRv+LV9MX5uipxlwaOVQ5f@xxxxxxxxxxxxxxxx
> https://lists.sourceforge.net/lists/listinfo/amportal-users
> 
> 
> End of Amportal-users Digest 
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