[Amportal-users] A few questions

OK, Finally got Trixbox 1.1.1 going with FreePBX 2.1.1.

Lots of reading later and knowing there’s more I don’t know than what
I do know, these are the answers I’m stuck on at present. If this is
the wrong forum for these questions please advise.

  1. Recording Calls - *1 does not seem to do anything. Is there
    something that needs to be done to activate it? The extensions are
    set to “On Demand.”

  2. Recording in general - If the voice mail recordings seem muddy and
    not very loud. What component should I look to. PC hardware with a
    clone X101P for PSTN connections and a PAP2 to feed my home’s phone
    lines. The tx/rx values don’t reflect changes made in zapata.conf
    when running zttool. Is the card bad?

  3. Pattern matching - I took the Blacklist module and used part of it
    to create a White List that filters our evening callers to only those
    on the list. Is there any way to do pattern matching so that ranges
    of numbers can be evaluated in one pass using Asterisk’s built-in
    database?

  4. Backup/Restore - Does this work on FreePBX 2.1.1 as intended? I’ve
    read conflicting reports.

  5. Distinctive Ring - Can it be done by Trunk to an extension
    connected to a PAP2? I’d like to know if a call is coming in is from
    a business line or a personal line. I used to subscribe to a virtual
    PBX that would alert me of an incoming business call by playing a
    couple of beeps prior to transferring the call. I then knew how to
    answer the call.

  6. This last one is not FreePBX specific but maybe someone knows the
    answer. At what point does a VoIP call to a landline leave the
    internet and get on phone company type circuits? I’m trying to
    determine if getting a VoIP provider that’s geographically close will
    increase the odds of getting better call quality, everything else
    being equal.

Thanks for reading.


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Ooops. I misread your question 1. More than likely that is caused by
Asterisk not detecting the *1. Make sure your phone has DTMF set to RFC
mode.


Michael Shuler, C.E.O.
OmniLEC And BitWise Communications, Inc.

[quote] -----Original Message-----
From: [email protected]
[mailto:[email protected]] On
Behalf Of Michael Shuler
Sent: Thursday, August 10, 2006 10:19 PM
To: [email protected]
Subject: Re: [Amportal-users] A few questions

  1. If during a barge from HUD you need to enable it under
    /etc/asterisk/extensions_hud.conf by changing the [barge] section to:
    [barge]
    ;
    ; support the ability of barging into calls
    ;
    exten => _1X.,1,MeetMe(20000000${EXTEN:1},qd,271721)
    exten => _2X.,1,MeetMe(20000000${EXTEN:1},aqds,271721)

  2. Most likely it’s the phone you are calling from but it
    could be some
    weird settings that you will have to play with for the rest
    of your life
    until you get lucky. Try a Grandstream GXP-2000 directly to
    it, that’s what
    we use and everything is crystal clear. We also gateway to
    the PSTN via a
    PRI which makes calls sound perfect (Assuming PRI or VoIP on
    the other end).

  3. Sorry, never tried that.

  4. Sorry, never tried that yet either.

  5. Once it reaches the VoIP carrier its up to them as to if
    they are going
    to drop it out directly to the PSTN from he gateway you came
    in on or if
    they are going to ship it around via VoIP, TDM or ATM. Most
    of the time it
    really wont matter. But every VoIP service provider will be
    using digital
    lines i.e. PRI to interface to the PSTN, hence pending any Internet
    flakeyness your call quality will be better than anything you try with
    analog lines at your home or business. As to how much, well
    that depends on
    how good your analog lines are.


Michael Shuler, C.E.O.
OmniLEC And BitWise Communications, Inc.

-----Original Message-----
From: [email protected]
[mailto:[email protected]] On
Behalf Of Carlos Leal
Sent: Thursday, August 10, 2006 9:51 PM
To: [email protected]
Subject: [Amportal-users] A few questions

OK, Finally got Trixbox 1.1.1 going with FreePBX 2.1.1.

Lots of reading later and knowing there’s more I don’t know
than what
I do know, these are the answers I’m stuck on at present.
If this is
the wrong forum for these questions please advise.

  1. Recording Calls - *1 does not seem to do anything. Is there
    something that needs to be done to activate it? The extensions are
    set to “On Demand.”

  2. Recording in general - If the voice mail recordings seem
    muddy and
    not very loud. What component should I look to. PC hardware with a
    clone X101P for PSTN connections and a PAP2 to feed my
    home’s phone
    lines. The tx/rx values don’t reflect changes made in zapata.conf
    when running zttool. Is the card bad?

  3. Pattern matching - I took the Blacklist module and used
    part of it
    to create a White List that filters our evening callers to
    only those
    on the list. Is there any way to do pattern matching so
    that ranges
    of numbers can be evaluated in one pass using Asterisk’s built-in
    database?

  4. Backup/Restore - Does this work on FreePBX 2.1.1 as
    intended? I’ve
    read conflicting reports.

  5. Distinctive Ring - Can it be done by Trunk to an extension
    connected to a PAP2? I’d like to know if a call is coming
    in is from
    a business line or a personal line. I used to subscribe to
    a virtual
    PBX that would alert me of an incoming business call by playing a
    couple of beeps prior to transferring the call. I then knew how to
    answer the call.

  6. This last one is not FreePBX specific but maybe someone
    knows the
    answer. At what point does a VoIP call to a landline leave the
    internet and get on phone company type circuits? I’m trying to
    determine if getting a VoIP provider that’s geographically
    close will
    increase the odds of getting better call quality, everything else
    being equal.

Thanks for reading.



Using Tomcat but need to do more? Need to support web
services, security?
Get stuff done quickly with pre-integrated technology to make
your job easier
Download IBM WebSphere Application Server v.1.0.1 based on
Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&
dat=121642


Amportal-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/amportal-users



Using Tomcat but need to do more? Need to support web
services, security?
Get stuff done quickly with pre-integrated technology to make
your job easier
Download IBM WebSphere Application Server v.1.0.1 based on
Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&
dat=121642


Amportal-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/amportal-users

[/quote]


Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with pre-integrated technology to make your job easier
Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642


Amportal-users mailing list
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  1. If during a barge from HUD you need to enable it under
    /etc/asterisk/extensions_hud.conf by changing the [barge] section to:
    [barge]
    ;
    ; support the ability of barging into calls
    ;
    exten => _1X.,1,MeetMe(20000000${EXTEN:1},qd,271721)
    exten => _2X.,1,MeetMe(20000000${EXTEN:1},aqds,271721)

  2. Most likely it’s the phone you are calling from but it could be some
    weird settings that you will have to play with for the rest of your life
    until you get lucky. Try a Grandstream GXP-2000 directly to it, that’s what
    we use and everything is crystal clear. We also gateway to the PSTN via a
    PRI which makes calls sound perfect (Assuming PRI or VoIP on the other end).

  3. Sorry, never tried that.

  4. Sorry, never tried that yet either.

  5. Once it reaches the VoIP carrier its up to them as to if they are going
    to drop it out directly to the PSTN from he gateway you came in on or if
    they are going to ship it around via VoIP, TDM or ATM. Most of the time it
    really wont matter. But every VoIP service provider will be using digital
    lines i.e. PRI to interface to the PSTN, hence pending any Internet
    flakeyness your call quality will be better than anything you try with
    analog lines at your home or business. As to how much, well that depends on
    how good your analog lines are.


Michael Shuler, C.E.O.
OmniLEC And BitWise Communications, Inc.

[quote] -----Original Message-----
From: [email protected]
[mailto:[email protected]] On
Behalf Of Carlos Leal
Sent: Thursday, August 10, 2006 9:51 PM
To: [email protected]
Subject: [Amportal-users] A few questions

OK, Finally got Trixbox 1.1.1 going with FreePBX 2.1.1.

Lots of reading later and knowing there’s more I don’t know
than what
I do know, these are the answers I’m stuck on at present. If this is
the wrong forum for these questions please advise.

  1. Recording Calls - *1 does not seem to do anything. Is there
    something that needs to be done to activate it? The extensions are
    set to “On Demand.”

  2. Recording in general - If the voice mail recordings seem
    muddy and
    not very loud. What component should I look to. PC hardware with a
    clone X101P for PSTN connections and a PAP2 to feed my home’s phone
    lines. The tx/rx values don’t reflect changes made in zapata.conf
    when running zttool. Is the card bad?

  3. Pattern matching - I took the Blacklist module and used
    part of it
    to create a White List that filters our evening callers to
    only those
    on the list. Is there any way to do pattern matching so that ranges
    of numbers can be evaluated in one pass using Asterisk’s built-in
    database?

  4. Backup/Restore - Does this work on FreePBX 2.1.1 as
    intended? I’ve
    read conflicting reports.

  5. Distinctive Ring - Can it be done by Trunk to an extension
    connected to a PAP2? I’d like to know if a call is coming in is from
    a business line or a personal line. I used to subscribe to a virtual
    PBX that would alert me of an incoming business call by playing a
    couple of beeps prior to transferring the call. I then knew how to
    answer the call.

  6. This last one is not FreePBX specific but maybe someone knows the
    answer. At what point does a VoIP call to a landline leave the
    internet and get on phone company type circuits? I’m trying to
    determine if getting a VoIP provider that’s geographically
    close will
    increase the odds of getting better call quality, everything else
    being equal.

Thanks for reading.



Using Tomcat but need to do more? Need to support web
services, security?
Get stuff done quickly with pre-integrated technology to make
your job easier
Download IBM WebSphere Application Server v.1.0.1 based on
Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&
dat=121642


Amportal-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/amportal-users

[/quote]


Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with pre-integrated technology to make your job easier
Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642


Amportal-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/amportal-users

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