Re: [Asterisk-Users] Voicemail -> new feature request
I'll throw in a few requests as well- A "pause" feature. The ability to mark a recording "urgent". The ability to change the prompt features around and edit voicemail prompts, recording abilities, while retaining defaults and customizations for different extensions. I'm still studying AGI's and the record() app as well as the RECORD_FILE agi command but am very new to programming. Anyone who has had success with modyfying voicemail in any way please share your stories. Thanks! From: "Matthew T. O'Connor" Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion CC: Asterisk Developers Mailing List Subject: Re: [Asterisk-Users] Voicemail -> new feature request Date: Fri, 14 Oct 2005 12:16:50 -0400 Kib Eki wrote: It really would be nice if each user is able to active/deactivate the mail forwarding of his voicemail via the VoiceMailMenu. Also, one simple thing. Is it possible to listen to my greetings without having to re-record them? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Modifying cmd VoicemailMain
I have also been looking for a way to customize voicemail (I want to add a "pause" feature and change the promps). I have come to the same conclusions as to where to do it, but have not yet created a solution. I have found this posting/forum which gives insight into modifying the "app_voicemail.c" file but does not directly address our issues, it's good info however: http://www.voipuser.org/forum_topic_2952.html Something tells me it will be a combination of modification, agi and dialplan. There is also another person with the similar goals here: http://sourceforge.net/forum/forum.php?thread_id=1363794&forum_id=420324 If you examine the app_voicemail.c file it refers to these links/people who contributed to different language translations and additions: * 12-16 - 2005 : Support for Greek added by InAccess Networks (work funded by HOL, www.hol.gr) *George Konstantoulakis <[EMAIL PROTECTED]> * 05-10 - 2005 : Support for Swedish and Norwegian added by Daniel Nylander, http://www.danielnylander.se/ * * 05-11 - 2005 : An option for maximum number of messsages per mailbox added by GDS Partners (www.gdspartners.com) *Stojan Sljivic <[EMAIL PROTECTED]> * * 07-11 - 2005 : An issue with voicemail synchronization has been fixed by GDS Partners (www.gdspartners.com) *Stojan Sljivic <[EMAIL PROTECTED]> And of course there is [EMAIL PROTECTED] (I'm not ready to bother him yet, although someone willing to spearhead a Japanese translation project might get some attention or at least a reference). I would love to hear about anyone's success in this area! And I will be sure to post my progress in these threads. F From: "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Modifying cmd VoicemailMain Date: Wed, 12 Oct 2005 18:18:26 -0700 On Thu, 2005-10-13 at 10:08 +0900, Kuniyoshi Murata wrote: > Andy Kuo writes: > > > Hi, > > Maybe you can record the sound file "vm-five.gsm" as "five hour" in > > Japanese, instead of just "five". > > AK > > I don't think you can do that. > Because that vm-five.gsm can be used as message number also (e.g. "message FIVE") > For the other changes I am starting to think that it will require either modifying the voicemail app or doing voicemail as an agi or dialplan setup. All 3 have some drawbacks, but would give you the ability to tweak everything exactly how you want it. As either an agi or dialplan setting you could use most of the voicemail app functionality if that is suitable (I dont know where the prompts are exactly that the original poster refered to). It may boil down to writing a complete voicemail system as an agi or modifying the voicemail app to get exactly what is wanted. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] parameters documentation
The people who have been documenting Asterisk have been working on a book for the last few months, it has been published by O'reilly (Asterisk-The Future of Telephony)and is just now finding it's way into the major bookstores, listed under Open-Source at Barns&Noble. While it will not answer everything asterisk can do, but its glossery and and appendix are very helpful for quick reference. If you have been following the Asterisk Documentation Project, some of it will be old hat, but I'm looking forward to replacing my huge stack of printouts with it. It gives a pretty good overview of VOIP, Networking, Telephony, etc. http://www.oreilly.com/catalog/asterisk/ From: "Steve Totaro" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [Asterisk-Users] parameters documentation Date: Wed, 12 Oct 2005 11:17:01 -0400 There is plenty of documentation online for both the 3com and *. You have to have good search skills I guess. 3com has the best knowledge base I have seen. http://knowledgebase.3com.com/ and there are tons of 3com dealers that can help. I think you may need to learn some basic networking before learning asterisk. NAT is a very basic concept in networking as well as ports such as 5060 (standard port for SIP). There is a very steep learning curve for asterisk and networking in general. If you want to learn it then you need to dig into the wiki and read all the posts that come across the user's list (well maybe not all of them). There are plenty of consultants that you can hire if you are not up to it. - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" ; <[EMAIL PROTECTED]> Sent: Wednesday, October 12, 2005 10:34 AM Subject: Re: [Asterisk-Users] parameters documentation > I come from a NBX100 > No documentation available. > 1 day it starts saying: "syslog full" and voicemail stop working > No one was able to tell me what was the meaning of that alert > . > 3COM NBX anyway is a good product, but the price is too high, especially 4 > years ago, and especially the price of the telephone is very high. > > Andrea > > > > > > "asterisk" > <[EMAIL PROTECTED] > echnologies.com> To > Sent by: "Asterisk Users Mailing List - > asterisk-users-bo Non-Commercial Discussion" > [EMAIL PROTECTED] > m.com cc > > Subject > 13/10/2005 16.13 Re: [Asterisk-Users] parameters >documentation > > Please respond to > Asterisk Users > Mailing List - > Non-Commercial > Discussion > <[EMAIL PROTECTED] > ists.digium.com> > > > > > > > "> I really hope this project will be implemented, without documentation > evrything is too hard" > > Not for the thousands of people that have figured it out. > > 3Com NBX might be more your speed and plenty of documentation. > > > > > Really strange answer. I am non used to search on playboy.com. > > > > Anyway, if you try to search > > insecure=very > > on www.voip-info.org, you find 742 links , a bit more for me. (I just > want > > to know what it means) > > > > Moreovere, the first 20 links are non accessible at all > > > > > http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecure&diff=6 > > > > > they speak about tiki-pagehistory.php, which appears not to exist. > > > > no other comments about this. > > > > > > I know about one project , "asterisk documentation project" > > > > http://www.asteriskdocs.org > > > > in its home page, the first line is > > > > > > > > > > > > Great software needs great documentation. > > > > > > > > > > I really hope this project will be implemented, without documentation > > evrything is too hard > > > > Andrea > > > > > > > > > > "Steve Totaro" > > <[EMAIL PROTECTED] > > echnologies.com> > To > > Sent by: "Asterisk Users Mailing List - > > asterisk-users-bo Non-Commercial Discussion" > > [EMAIL PROTECTED] > > m.com > cc > > > > > Subject > > 12/10/2005 14.53 Re: [Asterisk-Users] parameters > >documentation > > > > Please respond to > > Asterisk Users > > Mailing List - > > Non-Commercial > > Discussion > > <[EM
[Asterisk-Users] Anyone using Asterisk to take credit card payments?
I want to have customers make payments by keying in their cc#'s. I can see it's possible, I just want to know if anyone out there is doing this and what financial institutions are supporting Asterisk PBX's. So far I have found a few leads but would like to check here at the same time. Thank you! _N _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8
I don't know for sure, but most likely during the /usr/src/zaptel make &&
make install or during the /usr/src/asterisk make && make install. Messing
with the Makefiles in either one of those might allow you to do it with 2.6,
I tried, but do not have the skills yet. I also tried to manually create
the /dev/zap/ directory hoping that the ctl would load into it but it would
dissappear during restarts.
Someone else might have solved this with 2.6, possibly with Debian.
I would like to hear anyones success stories with AMP on a 2.6 Kernel
_N
From: Jachin Rupe <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] /dev/zap* is not showing up (gentoo,
portage,asterisk 1.0.8
Date: Tue, 6 Sep 2005 18:16:41 -0500
hi there
thanks a lot for the reply.
When exactly are the /dev/zap/ctl files supposed to be created?
I have been trying different things to see if they'll show up but known
when to expect them to show up would help a lot.
-jachin
On Sep 6, 2005, at 5:53 PM, FELIX E SKOWRONEK wrote:
I had this problem with White Box Enterprise Linux running the 2.6
kernel. When I went back to the 2.4 kernel it created the /dev/zap/ ctl
files. Still having other issues setting up AMP, but asterisk still
recommends the 2.4 kernel.
From: Jachin Rupe <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] /dev/zap* is not showing up (gentoo,
portage,asterisk 1.0.8)
Date: Tue, 6 Sep 2005 11:33:47 -0500
hi there
I'm trying to get asterisk going on gentoo 2005.1
I'm just getting my feet wet so I thought I would just stick with the
stable portage packages. Right now that's asterisk 1.0.8
I emerge asterisk with the following make.conf file:
CFLAGS="-O2 -mcpu=i686"
CHOST="i386-pc-linux-gnu"
CXXFLAGS="${CFLAGS}"
USE="-gtk -gnome -qt -kde -dvd alsa -cdr zaptel mmx mysql postgres
vmdbpostgres devfs26 pri"
So portage installed zaptel, pri everything else I should need.
Portage put the udev rules for zaptel here /etc/udev/rules.d/10-
zaptel.rules and it looks like this:
KERNEL="zapctl",NAME="zap/ctl", MODE="0660", GROUP="dialout"
KERNEL="zaptimer", NAME="zap/timer", MODE="0660", GROUP="dialout"
KERNEL="zapchannel",NAME="zap/channel", MODE="0660", GROUP="dialout"
KERNEL="zappseudo", NAME="zap/pseudo", MODE="0660", GROUP="dialout"
KERNEL="zap[0-9]*", NAME="zap/%n", MODE="0660", GROUP="dialout"
Here's what an lspci looks like:
:00:00.0 Host bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ ZX/ DX
Host bridge (rev 03)
:00:01.0 PCI bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ ZX/ DX
AGP bridge (rev 03)
:00:02.0 ISA bridge: Intel Corporation 82371AB/EB/MB PIIX4 ISA (rev
02)
:00:02.1 IDE interface: Intel Corporation 82371AB/EB/MB PIIX4 IDE
(rev 01)
:00:02.2 USB Controller: Intel Corporation 82371AB/EB/MB PIIX4 USB
(rev 01)
:00:02.3 Bridge: Intel Corporation 82371AB/EB/MB PIIX4 ACPI (rev 02)
:00:03.0 Ethernet controller: Intel Corporation 82557/8/9 [Ethernet
Pro 100] (rev 05)
:00:10.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
:00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
:01:01.0 VGA compatible controller: S3 Inc. Trio 64 3D (rev 01)
As you can see I have two X100P clones.
If I run #/etc/init.d/zaptel start
* Starting zaptel ...
Notice: Configuration file is /etc/zaptel.conf
line 143: Unable to open master device '/dev/zap/ ctl'
[ ok ]
running ztcfg -vv give me a similar message:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
Notice: Configuration file is /etc/zaptel.conf
line 122: Unable to open master device '/dev/zap/ctl'
Sure enough... if I look in dev there's no /dev/zap
This is what my /etc/zaptel.conf looks like:
fxsks=1-2
loadzone = us
defaultzone=us
It seems as though there's some step I'm missing but after looking
around for a while I have not been able to find it.
I'm wondering why /dev/zap isn't showing up?
I would be more than happy to supply any more information that I left
out.
If I figure anything else out on my own I'll be sure to post it.
thanks.
-jachin
___
--Bandwidth and
RE: [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage, asterisk 1.0.8
I had this problem with White Box Enterprise Linux running the 2.6 kernel.
When I went back to the 2.4 kernel it created the /dev/zap/ctl files. Still
having other issues setting up AMP, but asterisk still recommends the 2.4
kernel.
From: Jachin Rupe <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] /dev/zap* is not showing up (gentoo,
portage,asterisk 1.0.8)
Date: Tue, 6 Sep 2005 11:33:47 -0500
hi there
I'm trying to get asterisk going on gentoo 2005.1
I'm just getting my feet wet so I thought I would just stick with the
stable portage packages. Right now that's asterisk 1.0.8
I emerge asterisk with the following make.conf file:
CFLAGS="-O2 -mcpu=i686"
CHOST="i386-pc-linux-gnu"
CXXFLAGS="${CFLAGS}"
USE="-gtk -gnome -qt -kde -dvd alsa -cdr zaptel mmx mysql postgres
vmdbpostgres devfs26 pri"
So portage installed zaptel, pri everything else I should need.
Portage put the udev rules for zaptel here /etc/udev/rules.d/10-
zaptel.rules and it looks like this:
KERNEL="zapctl",NAME="zap/ctl", MODE="0660", GROUP="dialout"
KERNEL="zaptimer", NAME="zap/timer", MODE="0660", GROUP="dialout"
KERNEL="zapchannel",NAME="zap/channel", MODE="0660", GROUP="dialout"
KERNEL="zappseudo", NAME="zap/pseudo", MODE="0660", GROUP="dialout"
KERNEL="zap[0-9]*", NAME="zap/%n", MODE="0660", GROUP="dialout"
Here's what an lspci looks like:
:00:00.0 Host bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/ DX
Host bridge (rev 03)
:00:01.0 PCI bridge: Intel Corporation 440BX/ZX/DX - 82443BX/ZX/ DX AGP
bridge (rev 03)
:00:02.0 ISA bridge: Intel Corporation 82371AB/EB/MB PIIX4 ISA (rev
02)
:00:02.1 IDE interface: Intel Corporation 82371AB/EB/MB PIIX4 IDE (rev
01)
:00:02.2 USB Controller: Intel Corporation 82371AB/EB/MB PIIX4 USB
(rev 01)
:00:02.3 Bridge: Intel Corporation 82371AB/EB/MB PIIX4 ACPI (rev 02)
:00:03.0 Ethernet controller: Intel Corporation 82557/8/9 [Ethernet
Pro 100] (rev 05)
:00:10.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
:00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
:01:01.0 VGA compatible controller: S3 Inc. Trio 64 3D (rev 01)
As you can see I have two X100P clones.
If I run #/etc/init.d/zaptel start
* Starting zaptel ...
Notice: Configuration file is /etc/zaptel.conf
line 143: Unable to open master device '/dev/zap/ ctl'
[ ok ]
running ztcfg -vv give me a similar message:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
Notice: Configuration file is /etc/zaptel.conf
line 122: Unable to open master device '/dev/zap/ctl'
Sure enough... if I look in dev there's no /dev/zap
This is what my /etc/zaptel.conf looks like:
fxsks=1-2
loadzone = us
defaultzone=us
It seems as though there's some step I'm missing but after looking around
for a while I have not been able to find it.
I'm wondering why /dev/zap isn't showing up?
I would be more than happy to supply any more information that I left out.
If I figure anything else out on my own I'll be sure to post it.
thanks.
-jachin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_
Dont just search. Find. Check out the new MSN Search!
http://search.msn.click-url.com/go/onm00200636ave/direct/01/
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
