Re: [asterisk-users] asterisk-users Confbridge

2020-08-07 Thread Matthew Fredrickson 
Sorry about the trouble.  Unsubscribed that user from the mailing lists.

Matthew Fredrickson

On Fri, Aug 7, 2020 at 9:20 PM Elizabeth  wrote:
>
> I'm online on this site!
> So contact me in my profile:
> here
> --
> _
> -- 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

-- 
_
-- 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] asterisk-users Confbridge

2020-08-07 Thread Elizabeth 
Im online on this site!  So contact me in my profile:  
galleries.daswanitailors.com here
-- 
_
-- 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] Confbridge

2020-08-07 Thread Sam Basan 
John,

What you see it's how it should be.

The wait for admin means that all users join the conference room but the
conference is not started and they all should hear MOH.

When the admin will join then the conference will start and all will hear
the admin (or all others if they are not muted)

 

 

 

 

 

 



 

Office:  +972-77-6662000  E-mail:
 off...@bluebe.net 

Fax:   +972-77-6662020  Web:
 http://www.bluebe.net   

 



 

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of John T. Bittner
Sent: Saturday, August 8, 2020 12:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion‏

Subject: [asterisk-users] Confbridge

 

To all:


No matter what I try, I cannot get the system to wait for the admin to join.
It just dumps users into the bridge directly.

I do not have a pin for users, does that matter?

 

What am I missing?

 

Another issue the absolute timeout is not working ? … have recordings that
last for over 24 hours… and this should not happen…

All calls should hangup after 4 ?

 

Any ideas ?

 

Any help is much appreciated.

 

Thanks

 

This is my dialplan.

 

exten => s,1,Wait(1)

exten => s,n,Answer

exten => s,n,Set(TIMEOUT(absolute)=14400)

exten => s,n,NoOp(${CALLERID(name)})

exten => s,n,NoOp(${CALLERID(num)})

exten => s,n,NoOp()

exten => s,n,Playback(church) ; "Please hold while..."

exten => s,n,Set(CONFBRIDGE(user,announce_join_leave)=no)

exten => s,n,Set(CONFBRIDGE(user,startmuted)=yes)

exten => s,n,Set(CONFBRIDGE(user,template)=church)

exten => s,n,Set(CONFBRIDGE(user,marked)=no)

exten => s,n,Set(CONFBRIDGE(user,wait_marked)=yes)

exten => s,n,Set(CONFBRIDGE(user,end_marked)=yes)

exten => s,n,ConfBridge(xaccel)

exten => s,n,hangup

 

confbridge.conf

 

[general]

[church]

type=user

startmuted=yes

announce_join_leave=no

announce_user_count=no

wait_marked=yes

end_marked=yes

music_on_hold_when_empty=no

quiet=yes

;

[xaccel]

type=bridge

record_conference=yes

;

Then calling in I see this 

Conference Bridge Name   Users  Marked Locked Muted

 == == == =

xaccel1  0 No No

 

 

John Bittner

CTO



380 US Highway 46, Suite 500

Totowa, NJ 07512

Phone: 201.806.2602 x2405

Fax:   201.806.2604

Cell:   973.390.1090

  www.xaccel.net

 

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the
sender by reply e-mail and destroy all copies of the e-mail.

 

-- 
_
-- 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

[asterisk-users] Confbridge

2020-08-07 Thread John T. Bittner 
To all:

No matter what I try, I cannot get the system to wait for the admin to join. It 
just dumps users into the bridge directly.
I do not have a pin for users, does that matter?

What am I missing?

Another issue the absolute timeout is not working ? ... have recordings that 
last for over 24 hours... and this should not happen...
All calls should hangup after 4 ?

Any ideas ?

Any help is much appreciated.

Thanks

This is my dialplan.

exten => s,1,Wait(1)
exten => s,n,Answer
exten => s,n,Set(TIMEOUT(absolute)=14400)
exten => s,n,NoOp(${CALLERID(name)})
exten => s,n,NoOp(${CALLERID(num)})
exten => s,n,NoOp()
exten => s,n,Playback(church) ; "Please hold while..."
exten => s,n,Set(CONFBRIDGE(user,announce_join_leave)=no)
exten => s,n,Set(CONFBRIDGE(user,startmuted)=yes)
exten => s,n,Set(CONFBRIDGE(user,template)=church)
exten => s,n,Set(CONFBRIDGE(user,marked)=no)
exten => s,n,Set(CONFBRIDGE(user,wait_marked)=yes)
exten => s,n,Set(CONFBRIDGE(user,end_marked)=yes)
exten => s,n,ConfBridge(xaccel)
exten => s,n,hangup

confbridge.conf

[general]
[church]
type=user
startmuted=yes
announce_join_leave=no
announce_user_count=no
wait_marked=yes
end_marked=yes
music_on_hold_when_empty=no
quiet=yes
;
[xaccel]
type=bridge
record_conference=yes
;
Then calling in I see this
Conference Bridge Name   Users  Marked Locked Muted
 == == == =
xaccel1  0 No No


John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

-- 
_
-- 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] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?

2020-08-07 Thread Dan Cropp 
Thank you Jöran

I also figured out my problem with the caller id name/number.  In case anyone 
else encounters the caller id name issue, replace the spaces in the name with 
control sequence for a space %20

From: asterisk-users  On Behalf Of 
Jöran Vinzens
Sent: Friday, August 7, 2020 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] With ARI, is it possible to create (originate) a 
call and pass both the caller id name and number?

Hi Dan,

as far as PPI and PAI Header, we use the channel Vars in order to do that. In 
Latest Asterisk you can set Channel vars within the create command in the Body. 
Just set the PJSIP(add,PAI) as you would do it in Dialplan.
https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/
BR
Jöran

On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp 
mailto:d...@amtelco.com>> wrote:
An additional follow-up question, if I need to set the P-Asserted-Identity on 
the create (originate), is there a way to do this with ARI?

From: asterisk-users 
mailto:asterisk-users-boun...@lists.digium.com>>
 On Behalf Of Dan Cropp
Sent: Friday, August 7, 2020 11:51 AM
To: 'asterisk-users@lists.digium.com' 
mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users] With ARI, is it possible to create (originate) a call 
and pass both the caller id name and number?

I’m trying to transition from AMI to ARI.

Running into a small hiccup when I try to create (originate a call) with the 
caller id name and number

I can pass the Name and Number if the name has no spaces in it and it shows up 
in my PhonerLite application.

curl -v -u asterisk:asterisk -X POST 
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291>

However, when the caller id name has a space in it, I can’t figure out how to 
pass the name and number successfully.  The following only displays asterisk 
for the number and Dan for the name

curl -v -u asterisk:asterisk -X POST 
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan
 Cropp<291>

Here is an example of how we do this with AMI successfully.
Action: Originate
ActionID: S40
Channel: PJSIP/1003@1003
Exten: createcall
Context: IS
Priority: 1
Timeout: 6
CallerID: Dan Cropp <291>
Variable: 
CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2
Async: true

Dan
--
_
-- 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


--

Jöran Vinzens - vinz...@sipgate.de
Telefon: +49 211-63 55 56-21
Telefax: +49 211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391

www.sipgate.de - 
www.sipgate.co.uk
-- 
_
-- 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] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?

2020-08-07 Thread Jöran Vinzens 
Hi Dan,

as far as PPI and PAI Header, we use the channel Vars in order to do that.
In Latest Asterisk you can set Channel vars within the create command in
the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan.
https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/
BR
Jöran

On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp  wrote:

> An additional follow-up question, if I need to set the P-Asserted-Identity
> on the create (originate), is there a way to do this with ARI?
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *Dan Cropp
> *Sent:* Friday, August 7, 2020 11:51 AM
> *To:* 'asterisk-users@lists.digium.com' 
> *Subject:* [asterisk-users] With ARI, is it possible to create
> (originate) a call and pass both the caller id name and number?
>
>
>
> I’m trying to transition from AMI to ARI.
>
>
>
> Running into a small hiccup when I try to create (originate a call) with
> the caller id name and number
>
>
>
> I can pass the Name and Number if the name has no spaces in it and it
> shows up in my PhonerLite application.
>
>
>
> curl -v -u asterisk:asterisk -X POST
> http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291
> >
>
>
>
> However, when the caller id name has a space in it, I can’t figure out how
> to pass the name and number successfully.  The following only displays
> asterisk for the number and Dan for the name
>
>
>
> curl -v -u asterisk:asterisk -X POST
> http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan
> Cropp<291>
>
>
>
> Here is an example of how we do this with AMI successfully.
>
> Action: Originate
>
> ActionID: S40
>
> Channel: PJSIP/1003@1003
>
> Exten: createcall
>
> Context: IS
>
> Priority: 1
>
> Timeout: 6
>
> CallerID: Dan Cropp <291>
>
> Variable:
> CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2
>
> Async: true
>
>
>
> Dan
> --
> _
> -- 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



-- 

Jöran Vinzens - vinz...@sipgate.de
Telefon: +49 211-63 55 56-21
Telefax: +49 211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391

www.sipgate.de - www.sipgate.co.uk
-- 
_
-- 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] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?

2020-08-07 Thread Dan Cropp 
An additional follow-up question, if I need to set the P-Asserted-Identity on 
the create (originate), is there a way to do this with ARI?

From: asterisk-users  On Behalf Of Dan 
Cropp
Sent: Friday, August 7, 2020 11:51 AM
To: 'asterisk-users@lists.digium.com' 
Subject: [asterisk-users] With ARI, is it possible to create (originate) a call 
and pass both the caller id name and number?

I'm trying to transition from AMI to ARI.

Running into a small hiccup when I try to create (originate a call) with the 
caller id name and number

I can pass the Name and Number if the name has no spaces in it and it shows up 
in my PhonerLite application.

curl -v -u asterisk:asterisk -X POST 
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291>

However, when the caller id name has a space in it, I can't figure out how to 
pass the name and number successfully.  The following only displays asterisk 
for the number and Dan for the name

curl -v -u asterisk:asterisk -X POST 
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan
 Cropp<291>

Here is an example of how we do this with AMI successfully.
Action: Originate
ActionID: S40
Channel: PJSIP/1003@1003
Exten: createcall
Context: IS
Priority: 1
Timeout: 6
CallerID: Dan Cropp <291>
Variable: 
CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2
Async: true

Dan
-- 
_
-- 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

[asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?

2020-08-07 Thread Dan Cropp 
I'm trying to transition from AMI to ARI.

Running into a small hiccup when I try to create (originate a call) with the 
caller id name and number

I can pass the Name and Number if the name has no spaces in it and it shows up 
in my PhonerLite application.

curl -v -u asterisk:asterisk -X POST 
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291>

However, when the caller id name has a space in it, I can't figure out how to 
pass the name and number successfully.  The following only displays asterisk 
for the number and Dan for the name

curl -v -u asterisk:asterisk -X POST 
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan
 Cropp<291>

Here is an example of how we do this with AMI successfully.
Action: Originate
ActionID: S40
Channel: PJSIP/1003@1003
Exten: createcall
Context: IS
Priority: 1
Timeout: 6
CallerID: Dan Cropp <291>
Variable: 
CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2
Async: true

Dan
-- 
_
-- 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] One way audio on outgoing calls

2020-08-07 Thread Administrator 

Hi Carlos

Le 07/08/2020 à 06:33, Carlos Chavez a écrit :
    I am having a strange problem with a new provider.  We already 
have a couple SIP trunks working fine.  We are trying a new provider 
but we are having one way audio problems with outgoing calls. Incoming 
calls do have two way audio, only outgoing calls have this problem.  I 
do not see anything odd with a packet capture and using PJSIP history 
to check.  The provider says that on outgoing calls the get random 
characters instead of the media port for RTP.


    We are using Asterisk 16.12.0 with PJSIP.  The server is behind 
NAT so we have external_media_address and external_signaling_address 
set to the public IP and all relevant ports are forwarded to the 
Asterisk server.  The other SIP trunks work fine, only this new 
provider has a problem and only for outgoing calls.


    An rtp set debug on shows only outgoing packets to the media 
address but no incoming packets.  Why would there be a difference that 
makes it work on incoming calls but not on outgoing?


We faced this problem and it was a firewall issue on our side. But if 
you say that your provider doesn't get the RTP, I understand that they 
can't return anything. RTP ports ?


Cheers

--
Daniel

--
_
-- 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