[asterisk-users] Yealink VC200?
Hey, I'm thinking about buying a couple of Yealink VC200s for some of our teams, but I haven't been able to find much, if any, information about the VC200 support for Asterisk and other video conferencing solutions (Jitsi Meet). Does anyone know if this kit works okay with Asterisk etc? Cheers, Andrew -- Andrew Ruthven, Wellington, New Zealand MIITP At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Cloud : https://catalystcloud.nz GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2020: https://lca2020.linux.org.au/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference Call like conference done by mobile!
On Wed, 2016-10-05 at 17:34 +0530, Mandar Khire wrote: > hi, > I trying to solve one scenario:- > As I can make call from mobile phone to my friend1. As he accept it, > I put him on hold, & dial friend2. > As he also accept it, I put him on hold & follow same procedure till > friend6. > The I click on 'Merge call' & I can talk to all 6 friends at a time & > they can talk each other. > Can I write This scene by dialplan?How? > I used Confbridge but its different type of conference. > Need help. > Thanks. Hi Mandar, Check out the "addcaller" stuff here: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration Essentially you'd have a dialplan where you can call another number which is then added to the confbridge. Cheers, Andrew -- Andrew Ruthven, Wellington, New Zealand MIITP, CITPNZ At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Card : http://qr.catalyst.net.nz/907675e1 Cloud : NZs only real cloud - https://catalyst.net.nz/cloud GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2017: The Future of Open Source, Hobart, AU - http://linux.conf.au -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones when one is unregistered
On Thu, 2016-09-01 at 10:36 -0700, Dave Platt wrote: > > > Things can become more complicated in a couple of situations: > > (1) If one of the SIP users you specify isn't actually a SIP > endpoint device, but is a SIP identity on another system (PBX > or VoIP provider or etc.), then you really don't have any control > over how that endpoint would handle situations where the called > user isn't available. The endpoint might answer with *its* > voicemail, immediately. > > (2) If you were to dial a Local/ destination rather than a SIP/ > destination, then that dialing operation *is* run back through > your dialplan, and it might divert the call to voicemail > instantly. Another option is what I've had happen recently. I have my main number dial all the phones in my house, including an old Cisco 7905 that on busy or no answer would send back a 302 redirect to extension 8000 - VoiceMail. To make matters worse inbound callers would be dumped into VoiceMail as though they'd entered it from internally, rather than external. While I tried various different ways on the Cisco to stop that behaviour, I found the only solution was to tell the Dial() command to ignore the 302 by adding the i flag. Problem solved. Cheers, Andrew -- Andrew Ruthven, Wellington, New Zealand MIITP, ITCP At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Cloud : NZs only real cloud - https://catalyst.net.nz/cloud GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2017: The Future of Open Source, Hobart, AU - http://linux.conf.au -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail - Allow * for only some users
Hi John,
Ah ha! Excellent. That works.
Now for a further tweak, in my stdexten I set voicemail_option with
with b or u, as appropriate and use ${voicemail_option) instead of
option in the call to Voicemail below so the correct prompt is used.
Thank you!
On Thu, 2016-07-21 at 14:53 -0700, John Kiniston wrote:
> I think you almost have it.
>
> In your vmfwd context have a wildcard match that sends the caller
> back to the originating voicemail and then define specific extensions
> that are allowed to forward.
>
>
> [vmfwd]
> exten => _,1,Voicemail(box@context,option)
> same => n,Hangup
>
> ; Andrew Ruthven
> exten => 7231,1,Set(CALLERID(number)=yyy)
> same => n,Goto(pstn,xxx,1)
>
> On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthven <andrew.ruthven@catal
> yst.net.nz> wrote:
> > Hey,
> >
> > I have free calling to between DDIs and cellphones on our group
> > plan. I
> > figure it'd be nice to allow staff with those cellphones to be able
> > to
> > forward callers to their VoiceMail to their cellphones using the *
> > feature.
> >
> > I have a standard extension macro that has VoiceMail support.
> > So far I've done this by duplicating the standard extension macro,
> > and
> > adding this rule (where ARG1 is the extension):
> >
> > exten => a,1,Goto(vmfwd,${ARG1},1)
> >
> > Then in the vmfwd context I have rules like this (I need to set the
> > CALLERID(number) so our SIP provider accepts the call):
> >
> > ; Andrew Ruthven
> > exten => 7231,1,Set(CALLERID(number)=yyy)
> > exten => 7231,n,Goto(pstn,xxx,1)
> >
> > Which is working nicely. But, I thought, can I simplify this and
> > just
> > have one macro?
> >
> > So I've tried doing the following to fold it into my standard
> > extension
> > macro:
> >
> > 1) Tried using a/_7231 but that didn't match (well, it was worth a
> > try)
> > 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my
> > extension,
> > but if I disable the 7231 rules in vmfwd, I get:
> >
> > [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
> > __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to
> > invalid
> > extension but no invalid handler:
> > context,exten,priority=vmfwd,7231,1
> >
> > and the call hangs up, not a very nice user experience.
> >
> > The second option could work, as long as the user lands back into
> > VoiceMail if there is no valid extension. I thought about using
> > GoSub,
> > but how do I get the caller back into VoiceMail?
> >
> > I've done a bunch of searching for this, but haven't found any
> > general
> > solutions. Is it possible to do what I'm trying to achieve, or is
> > there
> > a better approach?
> >
> > This is Asterisk 11.13.
> >
> > Cheers,
> > Andrew
> >
> > --
> >
> > Andrew Ruthven, Wellington, New Zealand
> > MIITP, CITPNZ
> >
> > At work: andrew.ruth...@catalyst.net.nz
> > At home: and...@etc.gen.nz
> > Card : http://qr.catalyst.net.nz/907675e1
> > Cloud : NZs only real cloud - https://catalyst.net.nz/cloud
> > GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863
> > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
> >
> >
> >
> >
> >
> > --
> > ___
> > __
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com
> > --
> > New to Asterisk? Join us for a live introductory webinar every
> > Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
--
Andrew Ruthven, Wellington, New Zealand
MIITP, CITPNZ
At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Card : http://qr.catalyst.net.nz/907675e1
Cloud : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail - Allow * for only some users
Hey,
I have free calling to between DDIs and cellphones on our group plan. I
figure it'd be nice to allow staff with those cellphones to be able to
forward callers to their VoiceMail to their cellphones using the *
feature.
I have a standard extension macro that has VoiceMail support.
So far I've done this by duplicating the standard extension macro, and
adding this rule (where ARG1 is the extension):
exten => a,1,Goto(vmfwd,${ARG1},1)
Then in the vmfwd context I have rules like this (I need to set the
CALLERID(number) so our SIP provider accepts the call):
; Andrew Ruthven
exten => 7231,1,Set(CALLERID(number)=yyy)
exten => 7231,n,Goto(pstn,xxx,1)
Which is working nicely. But, I thought, can I simplify this and just
have one macro?
So I've tried doing the following to fold it into my standard extension
macro:
1) Tried using a/_7231 but that didn't match (well, it was worth a try)
2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my extension,
but if I disable the 7231 rules in vmfwd, I get:
[2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
__ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to invalid
extension but no invalid handler: context,exten,priority=vmfwd,7231,1
and the call hangs up, not a very nice user experience.
The second option could work, as long as the user lands back into
VoiceMail if there is no valid extension. I thought about using GoSub,
but how do I get the caller back into VoiceMail?
I've done a bunch of searching for this, but haven't found any general
solutions. Is it possible to do what I'm trying to achieve, or is there
a better approach?
This is Asterisk 11.13.
Cheers,
Andrew
--
Andrew Ruthven, Wellington, New Zealand
MIITP, CITPNZ
At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Card : http://qr.catalyst.net.nz/907675e1
Cloud : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail using object storage?
I'd say using s3fs (or similar) is an approach, but if VoiceMail had support baked into it for S3, then the integration would be better. I'll look into using one the FUSE based approaches as a stop-gap measure. ;) On Tue, 2016-02-16 at 13:12 +0100, Olivier wrote: > Isn't the purpose of s3fs-like addons (see [1]) to let S3 buckets be > mounted on Linux and thus allow any application like Asterisk make > use of it ? > > [1] https://github.com/s3fs-fuse/s3fs-fuse > > 2016-02-16 1:05 GMT+01:00 Andrew Ruthven <andrew.ruth...@catalyst.net > .nz>: > > Hey, > > > > I've found a bit of chatter about people using hacks to copy > > voicemail > > messages into object storage (like S3) after they've been recorded. > > But > > I was wondering if any work has been done on the VoiceMail app to > > actually store and retrieve messages to/from an object store? > > > > Cheers, > > Andrew > > -- > > Andrew Ruthven, Wellington, New Zealand > > MIITP, ITCP > > > > At work: andrew.ruth...@catalyst.net.nz > > At home: and...@etc.gen.nz > > Cloud : NZs only real cloud - https://catalyst.net.nz/cloud > > GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 > > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org > > > > > > > > -- > > ___ > > __ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > > -- > > New to Asterisk? Join us for a live introductory webinar every > > Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrew Ruthven, Wellington, New Zealand MIITP, ITCP At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Cloud : NZs only real cloud - https://catalyst.net.nz/cloud GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail using object storage?
Hey, I've found a bit of chatter about people using hacks to copy voicemail messages into object storage (like S3) after they've been recorded. But I was wondering if any work has been done on the VoiceMail app to actually store and retrieve messages to/from an object store? Cheers, Andrew -- Andrew Ruthven, Wellington, New Zealand MIITP, ITCP At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz Cloud : NZs only real cloud - https://catalyst.net.nz/cloud GPG fpr: C603 FC4E 600F 1CEC D1C8 D97C 4B53 D931 E4D3 E863 LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPv6 support?
On Tue, 2009-04-28 at 00:06 +0200, Hans Witvliet wrote: Sometime ago i got this status from _the_ guru... Russell replied, referencing 1.6.2... (but other code might get in-the-way) Thanks! From: Russell Bryant russ...@digium.com There has been progress. It is not yet merged into the main tree, though. I would expect it to go in within the first few releases of 1.6 ... Can anyone point me to where the IPv6 development is currently happening, I'd be more than happy to start playing with it. Cheers! -- Andrew Ruthven, Wellington, New Zealand At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz GPG fpr: 34CA 12A3 C6F8 B156 72C2 D0D7 D286 CE0C 0C62 B791 Follow the Signs, visit Wellington! Linux.conf.au 2010 http://www.lca2010.org.nz signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IPv6 support?
Hey, Just wondering if anyone can let me know what the status of IPv6 support for Asterisk is currently. I see that the branch where development was happening has gone away. I was trying: http://svn.digium.com/svn/asterisk/team/blanchet/v6 Has this branched moved to somewhere else? Cheers! -- Andrew Ruthven, Wellington, New Zealand At work: andrew.ruth...@catalyst.net.nz At home: and...@etc.gen.nz GPG fpr: 34CA 12A3 C6F8 B156 72C2 D0D7 D286 CE0C 0C62 B791 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Proxy - Still required?
Hi, With more recent version of v1.2 and with v1.4 are things like the AstManProxy still recommended if you want to have a bunch of applications talking directly to Asterisk? Cheers! -- Andrew Ruthven, Wellington, New Zealand At work: [EMAIL PROTECTED] At home: [EMAIL PROTECTED] GPG fpr: 34CA 12A3 C6F8 B156 72C2 D0D7 D286 CE0C 0C62 B791 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hospitals using Asterisk?
Hello, The IT folks at a hospital in New Zealand have approached us about deploying Asterisk, but they would like to talk to people at other hospitals that have already done this. If anyone works at a hospital that has deployed Asterisk, or deployed Asterisk at a hospital would you please get in touch with me? Either via email, or I'm currently at linux.conf.au in Sydney if you're here. Thanks! -- Andrew Ruthven, Wellington, New Zealand At work: [EMAIL PROTECTED] At home: [EMAIL PROTECTED] GPG fpr: 34CA 12A3 C6F8 B156 72C2 D0D7 D286 CE0C 0C62 B791 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
