Re: [asterisk-users] How to enable IM over the asterisk server

[Kristof Van Den Ouweland]

Hi Thyda,

I think you should see these as two individual systems. (I'm not an expert so 
just thinking out loud).

Since you mention that you did a SIP mapping on Openfire, may I assume that you 
have the Asterisk IM plugin?

In case of yes:
Yes, there is a plugin between OpenFire and Asterisk but it is not actively 
developed anymore since 2006
http://www.igniterealtime.org/projects/asterisk/

So I don't think the plugin is really realiable anymore  on current versions.

--

I consider them as 2 separate systems which have to work on their own. 
Unfortunatly this means that every softphone has 2 accounts: one is SIP to 
Asterisk, one is XMPP to Openfire.

That way our users are able to call internal/external using Asterisk, but do IM 
and internal calling via Openfire. (They can choose which source they take)

Openfire is connected to our AD so our users just can logon with their Windows 
credentials.

Unfortunatly, if you want a real production connection between Asterisk and 
Openfire, I'm unable to assist since I don't have the knowledge of it.
sorry

Hope this helps a bit.
kristof
>>> Thyda ENG  7/07/2015 11:28 >>>
Actually, I am using the openfire and I create two users with the SIP mapping 
on the openfire to the asterisk server. I can register one user with the 
openfire client(Spark) and yes it is connect to asterisk SIP also. But with the 
other one user, I register it with the SIP client(Zoiper/ or Linphone) and then 
I can make the call over these two SIP but they cannot reach the chat. I wonder 
what should I config between openfire and asterisk to enable chat over these 
two sip clients ?
I am waiting for your reply, Thank.

Thyda

On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland 
 wrote:


Good morning Thyda;

Perhaps somebody has a solution for using it on Asterisk itself but after some 
trying I added the Openfire server as a IM server.

I was a bit afraid that 'if' I got it working properly we had to maintain it 
and off course had to troubleshoot it in case it didn't work anymore.

I've read something that you add a ams_msg context in extensions.conf but that 
didn't work for me unfortunaly. It did work for SIP Messages on phones but not 
for IM.

I found Openfire easier to configure and it added a full integration with our 
LDAP which allowed single sign so that users could use the same password and 
log on automatically with the Jitsi client.

But if you have some specific questions, I will be glad to answer.

//Kristof
>>> Thyda ENG  7/07/2015 6:07 >>>
I am currently, I create the VOIP server which enable the user to make the call 
over the asterisk server, Additionally now I want the user to be able to chat 
to each other too.
I found some suggestion of using the openfire with asterisk but not much said 
on it, Anyway could you please share me how can I config the IM server over 
asterisk?

I am waiting for your reply,

Thyda

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Re: [asterisk-users] DTMF issue

[Tom Peters]

You probably have to reload asrerisk  after making the change. 

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/7/2015 3:53 PM >>>
Ah I see, in theory it's possible then. We don't have any IVRs or anything
which requires key presses, there isn't even voicemail on this particular
phone system so I don't think it will be too much of a problem.

I've also updated the firmware on the Cisco phones that have had the issue,
just to see if that solves the issue but as it's been going on for a while,
I'm not too confident it has.

Thanks,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport. 
See this discussion, which I found after I posted my first response: 
http://www.voip-info.org/wiki/view/Asterisk+DTMF 
Particularly this sentence: 
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in
sip.conf."

The big question for you is going to be, does your system need to recognize
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems
doing that? 



Thomas M. Peters | Systems Administrator | tpet...@mcts.org 
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org 
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26

Re: [asterisk-users] DTMF issue

[Jamie Rees]

Ah I see, in theory it's possible then. We don't have any IVRs or anything
which requires key presses, there isn't even voicemail on this particular
phone system so I don't think it will be too much of a problem.

I've also updated the firmware on the Cisco phones that have had the issue,
just to see if that solves the issue but as it's been going on for a while,
I'm not too confident it has.

Thanks,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue

In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport. 
See this discussion, which I found after I posted my first response: 
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Particularly this sentence: 
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in
sip.conf."

The big question for you is going to be, does your system need to recognize
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems
doing that? 



Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3'
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] c

[asterisk-users] Can I use ARI to update the builtin database, without executing the dial plan?

[Rodrigo Pimenta Carvalho]

Hi.

In my dial plan I can use the following commands to access and handle data from 
the builtin database.

DB
DB_DELETE
DB_EXISTS
DB_KEYS

It is OK for me. However, in my current project there will be an application 
responsible for recording new information in the builtin database. 
So, I need to know:

1.  Is it possible to access the builtin database, by means of ARI ?

2. If it is possible by ARI, where can I find a tutorial about it?

3. Can I do something like this?:

  My APP --sends data to a REST service> Asterisk REST 
Interfacethe data is put into the builtin 
database> The builtin database

That is, can I send data to the builtin database, using REST interfaces, but 
without executing any dial plan? If yes, my app will be able to update data 
without executing the dial plan.

Any hint will be very helpful!

Best Regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979(Brasil)
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Re: [asterisk-users] DTMF issue

[Tom Peters]

In my humble opinion, adjusting this setting will (for you) do nothing, since 
you don't use the dahdi channels for transport. 
See this discussion, which I found after I posted my first response: 
http://www.voip-info.org/wiki/view/Asterisk+DTMF 
Particularly this sentence: 
"Note: Asterisk 1.4 now also has the relaxdtmf= setting available in sip.conf."

The big question for you is going to be, does your system need to recognize 
inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems doing 
that? 



Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org 
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3'
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin
passthrough '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
received on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted
with begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
detected to have actual duration 78 on the wire, emulation will be triggered
on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has
duration 78 but want minimum 80, emulating on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation
of '3' queued on SIP/209

Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

[John Kiniston]

I don't see that the Authenticate application has return values for failure
cases or returns call control on a failure case.

Sorry I don't think you will be able to do what you want with it.

On Tue, Jul 7, 2015 at 12:22 PM, Motty Cruz  wrote:

>  Here is what i have,
> exten => _011X,1,Set(CALLERID(number)=factory2)
> exten => _011X,2,Authenticate(/home/asterisk/passwds.conf,m,3)
> exten => _011X,3,NoOp(user has been authenticated)
> exten => _011X,n,Dial(SIP/VoIPSP1/${EXTEN:1},80)
> exten => _011X,n,HangUp()
>
> I would like to add background music if authentication failed, then after
> 6 minutes hangup
>
> any ideas, suggestions?
>
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Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

[Motty Cruz]

Here is what i have,
exten => _011X,1,Set(CALLERID(number)=factory2)
exten => _011X,2,Authenticate(/home/asterisk/passwds.conf,m,3)
exten => _011X,3,NoOp(user has been authenticated)
exten => _011X,n,Dial(SIP/VoIPSP1/${EXTEN:1},80)
exten => _011X,n,HangUp()

I would like to add background music if authentication failed, then 
after 6 minutes hangup


any ideas, suggestions?

On 07/07/2015 09:09 AM, Motty Cruz wrote:

Hello,
I used this guide, it worked for me:
http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html

Thanks,

On 07/06/2015 04:54 PM, John Kiniston wrote:

The Authenticate application will do this for you.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate

You can either give it a single PIN to use for all calls, 
Authenticate using a value in the Asterisk Database, Or use a plain 
text file for the PIN's





On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz > wrote:


Hello All,

I will like to configure Asterisk to use PIN Code for all
outgoing international calls.

Also, any suggestions as to when should I prompt users for code
prior to dialing the number or after dialing the number?

can someone provide with a example on how to accomplish this
goal? I am a bit confuse by this :

http://forums.digium.com/viewtopic.php?p=130936&sid=707f657f7a61dfed55e4922304925091

Thanks for your help.


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A human being should be able to change a diaper, plan an invasion, 
butcher a hog, conn a ship, design a building, write a sonnet, 
balance accounts, build a wall, set a bone, comfort the dying, take 
orders, give orders, cooperate, act alone, solve equations, analyze a 
new problem, pitch manure, program a computer, cook a tasty meal, 
fight efficiently, die gallantly. Specialization is for insects.

---Heinlein






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Re: [asterisk-users] DTMF issue

[Jamie Rees]

Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from an upstream provider
(Gradwell.com).

Is it still possible to set this when using SIP trunks only and not physical
hardware? The box does have a Digium ISDN card but the ISDN is no longer
used.

My dahdi-channels.conf file looks stock: 

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63

Thanks again,
Jamie 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it happens at random on many external calls. If this
happens to you, especially on voice peaks (when the outside party said a
particularly loud syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a
broken DTMF tone and getting regenerated by your T1 or POTS card, or
Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by
using ...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone
from callers very well. They would dial 4 digits and in my logs, I'd see one
or two, maybe three. The autoattendant would tell them they had dialed an
invalid extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3'
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin
passthrough '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
received on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted
with begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
detected to have actual duration 78 on the wire, emulation will be triggered
on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has
duration 78 but want minimum 80, emulating on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation
of '3' queued on SIP/209-00021cac

 

Can someone please provide any tips? 

 

Thanks,

Jamie 



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Re: [asterisk-users] DTMF issue

[Tom Peters]

It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile 
phones but it happens at random on many external calls. If this happens to you, 
especially on voice peaks (when the outside party said a particularly loud 
syllable) then you probably have DTMF talk-off. 

I think it's caused by an audio tone mistakenly being interpreted at a broken 
DTMF tone and getting regenerated by your T1 or POTS card, or Asterisk itself. 

We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by using 
...
relaxdtmf=no
...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) 

Problem with that it that our autoattendant wasn't recognizing DTMF tone from 
callers very well. They would dial 4 digits and in my logs, I'd see one or two, 
maybe three. The autoattendant would tell them they had dialed an invalid 
extension. 

So we had to go back to relaxdtmf=yes on the dahdi channels in question. So 
problem_solved=no.

-T

Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org


>>> "Jamie Rees"  7/6/2015 4:53 PM >>>
Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance. 

I've tried changing the DTMF tones on the phones (512G's running firmware
7.5.5) from In-Band to every other possibility, but this hasn't helped at
all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals
do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they
couldn't really help much. I presume the issue is local to our phone system
but other than the logs below, have nothing to go on:

[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2'
received on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin
passthrough '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end '2'
received on SIP/sip-out-00021c6d, duration 200 ms
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end accepted
with begin '2' on SIP/sip-out-00021c6d
[2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF end passthrough
'2' on SIP/sip-out-00021c6d
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin '3'
received on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF begin
passthrough '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
received on SIP/209-00021cac, duration 90 ms
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end accepted
with begin '3' on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3'
detected to have actual duration 78 on the wire, emulation will be triggered
on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end '3' has
duration 78 but want minimum 80, emulating on SIP/209-00021cac
[2015-06-10 10:07:10] DTMF[5134][C-c5b6] channel.c: DTMF end emulation
of '3' queued on SIP/209-00021cac

 

Can someone please provide any tips? 

 

Thanks,

Jamie 



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[asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?

[Richard Kenner]

I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:

351 res = (int) *input * *value;

It's called from ast_frame_adjust_volume.

The frame looks like:

(gdb) print *f
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {
  id = AST_FORMAT_SLINEAR16, fattr = {format_attr = {
  0 }, rtp_marker_bit = 0 '\000'}}}, datalen = 0, 
  samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64, 
  src = 0x51623b0 "func_jitterbuffer interpolation", data = {ptr = 0x0, 
uint32 = 0, pad = "\000\000\000\000\000\000\000"}, delivery = {
tv_sec = 1436290187, tv_usec = 304285}, frame_list = {next = 0x0}, 
  flags = 0, ts = 0, len = 0, seqno = 0}

so datalen is 0 and samples nonzero.  ast_frame_adjust_volume, however,
iterates over samples, not datalen.  Is that correct?

What does it mean to have a packet with a zero datalen anyway?

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[asterisk-users] RES: What database should I use, for simple data storing? SQLite or the buitin one?

[Rodrigo Pimenta Carvalho]

Hi Antony.

Thank you for your replay. I have decided to use the builtin database, 
according to others help that I have kindly received in this discussion list. 
What I need is a simple solution, not a relational database one.

Best regards.

RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979   (Brasil)

De: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] em Nome de Антон Сацкий 
[satski...@gmail.com]
Enviado: terça-feira, 7 de julho de 2015 11:32
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] What database should I use, for simple data 
storing? SQLite or the buitin one?

Propose U to use Mysql

2015-07-07 17:26 GMT+03:00 Rodrigo Pimenta Carvalho 
mailto:pime...@inatel.br>>:


Hi.

I was studying about how to use databases in Asterisk, accessing it from the 
dial plan.
In my project, my dial plan will have to store simple data (ex: IP number, port 
number, device name, etc) in a persistent way, so that it will be possible to 
retrieve such information in future moments, still via dial plan.

For this case, I would like to know?

1. What is the best choice for storing and retrieving simple data , with dial 
plan instructions: SQLite or the builtin database option? Consider that I'm 
worried about installation, configuration and use difficulties.

2. Does Asterisk 13 come with SQLite ready for use or have I to install this 
database separately and configure it to be accessible in dial plan?

3. Where can I find tutorials about using SQLite or the builtin database for 
storing simple that?

P.S.: I'm not interested in storing CDR data.

Any hint will be very helpful!

Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979(Brasil)
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--
Best regards
Antony
моб (066) 919-75-33
моб (063) 656-43-40
satski...@gmail.com
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[asterisk-users] RES: Fwd: What database should I use, for simple data storing? SQLite or the buitin one?

[Rodrigo Pimenta Carvalho]

Hi.

Thank you for your instruction!
What I need is simplicity. That is, a simple solution (no relational data base) 
will fit very well now. In this case I will start investigating about how to 
use the Asterisk (version 13 or later) builtin database.

Is it SQLite?
How to access it via dial plan, etc?
What must I configure in my asterisk?
What must i install to use the builtin database?
Where to find a tutorial with explanations about such questions?

Best regards.

 


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979(Brasil)

De: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] em Nome de Technical Support 
[supp...@telium.ca]
Enviado: terça-feira, 7 de julho de 2015 11:51
Para: asterisk-users@lists.digium.com
Assunto: Re: [asterisk-users] Fwd:  What database should I use, for simple data 
storing? SQLite or the buitin one?

To some extent the answer depends on how you want to use it overall, and
what you already have installed.


We did something similar on a project where we created a simple app
accessible via AGI, and it stored/retrieved data to/from anXML file.  If
your access frequency is low enough that might be a good solution.  On
the other hand if you need complex query capability you should stay on
the SQL side.


  If you already have MySQL installed for other Asterisk features (eg:
CDR, or if you use FreePBX) then you might as well use that.

​


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[asterisk-users] RES: What database should I use, for simple data storing? SQLite or the buitin one?

[Rodrigo Pimenta Carvalho]

Hi John.

Thank you very much for you reply. It is exactly what I was needing to know. 
Now I will study about the use of SQLite + Asterisk, because MySQL will not be 
necessary in my solution. A relational database will not be necessary. 
I 'm needing simplicity.

Do you know where can I find a tutorial about accessing and using the asterisk 
builtin database, considering Asterisk 13 or later?

Any hint will be very helpful.

Thanks


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)

De: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] em Nome de Tech Support 
[aster...@voipbusiness.us]
Enviado: terça-feira, 7 de julho de 2015 11:58
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Assunto: Re: [asterisk-users] What database should I use,   for simple data 
storing? SQLite or the buitin one?

I believe that Asterisk 1.8 and older uses the BerkeleyDB for Asterisk's
internal database (AKA the Astdb) and in newer versions use SQLite. However,
the basic functionality is the same. Whether you use the Astdb or MySQL
really depends on what you want to do with it. The AstDB is not a relational
database like MySQL, it simply a key/value store. If you can get away with
that, and you need simplicity, then the AstDB is the way to go. If you need
MySQL you'll probably end up having to write AGI scripts to access it. Like
I said, it all depends on what your needs are.
Regards;
John

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo
Pimenta Carvalho
Sent: Tuesday, July 07, 2015 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] What database should I use, for simple data
storing? SQLite or the buitin one?



Hi.

I was studying about how to use databases in Asterisk, accessing it from the
dial plan.
In my project, my dial plan will have to store simple data (ex: IP number,
port number, device name, etc) in a persistent way, so that it will be
possible to retrieve such information in future moments, still via dial
plan.

For this case, I would like to know?

1. What is the best choice for storing and retrieving simple data , with
dial plan instructions: SQLite or the builtin database option? Consider that
I'm worried about installation, configuration and use difficulties.

2. Does Asterisk 13 come with SQLite ready for use or have I to install this
database separately and configure it to be accessible in dial plan?

3. Where can I find tutorials about using SQLite or the builtin database for
storing simple that?

P.S.: I'm not interested in storing CDR data.

Any hint will be very helpful!

Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979(Brasil)
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Re: [asterisk-users] Voicemail: saycid without prefix

[John Kiniston]

Nice!

I didn't know what dialing rules may apply to his location, Your code does
look like an improvement on mine tho.

I love the REGEX function.

Even better, if the first 4 digits are "0049", you could replace them with
> "0"
> as though it was an inland call:
>
> ExecIf(REGEX("^0049."
> ${CALLERID(NUM)})?Set(CALLERID(num)=0${CALLERID(NUM):4}))
>
> --
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

[Motty Cruz]

Hello,
I used this guide, it worked for me:
http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html

Thanks,

On 07/06/2015 04:54 PM, John Kiniston wrote:

The Authenticate application will do this for you.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate

You can either give it a single PIN to use for all calls, Authenticate 
using a value in the Asterisk Database, Or use a plain text file for 
the PIN's





On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz > wrote:


Hello All,

I will like to configure Asterisk to use PIN Code for all outgoing
international calls.

Also, any suggestions as to when should I prompt users for code
prior to dialing the number or after dialing the number?

can someone provide with a example on how to accomplish this goal?
I am a bit confuse by this :

http://forums.digium.com/viewtopic.php?p=130936&sid=707f657f7a61dfed55e4922304925091

Thanks for your help.


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--
A human being should be able to change a diaper, plan an invasion, 
butcher a hog, conn a ship, design a building, write a sonnet, balance 
accounts, build a wall, set a bone, comfort the dying, take orders, 
give orders, cooperate, act alone, solve equations, analyze a new 
problem, pitch manure, program a computer, cook a tasty meal, fight 
efficiently, die gallantly. Specialization is for insects.

---Heinlein




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Re: [asterisk-users] Issue call quality: Asterisk call quality on trunks

[Henry Fernandes]

It¹s not clear to me if you¹ve done troubleshooting to determine where the
quality issues are occurring.  Try testing outbound/external calls
separately from internal calls (i.e., calls that stay on your network and
don¹t go out over the trunk to the carrier).

If the problem is on internal calls, then I¹d say the quality issues are
caused by something local ‹ perhaps a networking issue or an issue with
virtualization.  If the problem is on internal calls, you should also test
calls without any codec conversion.

If the problem is on outbound/external calls only, then the issue might be
something on the LAN or ISP side.  As a first step, you could get a VoIP
Spear (voipspear.com) account and see if you can notice any problems with
that.  A second step would be to get packet captures.
-H

From:  Kristof Van Den Ouweland 
Reply-To:  Asterisk Users Mailing List - Non-Commercial Discussion

Date:  Monday, July 6, 2015 at 11:58 PM
To:  
Subject:  [asterisk-users] Issue call quality: Asterisk call quality on
trunks

Good afteroon all,

First of all: thanks for everybody who is willing to think this through with
me:

I'm having some issues regarding call quality between some calls. Let me try
to explain the situation first

We have a Asterisk 11.16 server based on the Xivo distribution. There are 2
servers running in cluster (Active Passive), both virtual with the following
config:
Quadcore CPU
8 GB ram
About 50Gb of diskspace which is used for about 15%
(Let's call this Asterisk cluster 001 for clarity)

The Asterisk server has a trunk to a cisco call manager which is on the same
site/LAN, and 4 trunks to other Asterisk servers (same distribution but
lower specs, name Asterisk cluster 002 and 003). These are all sites in our
WAN but they are geographically divided and connected via MPLS links.  Each
affiliate has a specific number range XXXYYY where XXX stands for the
affiliate and YYY is the extension of the users.
(Average bandwidth = 4Mpbs which has to be shared by applications. QoS
allows that VoIP is prioritized)

Now, the actual problem:

I've set my main codecs to G711 a-law, G7 222 (for cisco call manager) and
GSM as last. The GSM is set as primary for those trunks which don't have 4
Mbps of bandwidth available.

In most cases, trunk calling results in bad quality of conversations (a-law
is chosen as codec)  but or it is jitterish, or one party does not hear the
other party (complete silence) It could be that the second time they call,
everything is ok.

--

So a little ASCII map about the geographical setup:

Aff 1: [Asterisk cluster 001] <-- LAN trunk --> Cisco call manager
|
MPLS connection 20Mbps
|
|-->  MPLS Cloud<---> MPLS connection 2Mbps -->
[Asterisk cluster 002]
   | <>  MPLS
connection 4 Mbps --> [Asterisk cluster 003]

Calls between Cluster 001 <---> cluster 002 or 003 are potentially of bad
quality (sometimes ok but most of all jiterish)
Calls between Cluster 002 <---> cluster 003 are good

The bandwidth if cluster001 ( 20 Mbps) is used about 50% with peaks to 75%.

I've aslo actived the jitter buffer with a buffer of 200ms but this didn't
seem to do any good.

Does anybody have some hints how I can troubleshoot this?

Note: the Cisco calls to the other affiliaters over the same WAN don't have
issues but these are based on SCCP protocol.

Thanks in advance
Kristof




 
 
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Re: [asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one?

[Tech Support]

I believe that Asterisk 1.8 and older uses the BerkeleyDB for Asterisk's
internal database (AKA the Astdb) and in newer versions use SQLite. However,
the basic functionality is the same. Whether you use the Astdb or MySQL
really depends on what you want to do with it. The AstDB is not a relational
database like MySQL, it simply a key/value store. If you can get away with
that, and you need simplicity, then the AstDB is the way to go. If you need
MySQL you'll probably end up having to write AGI scripts to access it. Like
I said, it all depends on what your needs are.
Regards;
John 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo
Pimenta Carvalho
Sent: Tuesday, July 07, 2015 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] What database should I use, for simple data
storing? SQLite or the buitin one?



Hi.

I was studying about how to use databases in Asterisk, accessing it from the
dial plan. 
In my project, my dial plan will have to store simple data (ex: IP number,
port number, device name, etc) in a persistent way, so that it will be
possible to retrieve such information in future moments, still via dial
plan. 

For this case, I would like to know?

1. What is the best choice for storing and retrieving simple data , with
dial plan instructions: SQLite or the builtin database option? Consider that
I'm worried about installation, configuration and use difficulties.

2. Does Asterisk 13 come with SQLite ready for use or have I to install this
database separately and configure it to be accessible in dial plan?

3. Where can I find tutorials about using SQLite or the builtin database for
storing simple that?

P.S.: I'm not interested in storing CDR data.

Any hint will be very helpful!

Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979(Brasil)
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Re: [asterisk-users] Fwd: What database should I use, for simple data storing? SQLite or the buitin one?

[Technical Support]

To some extent the answer depends on how you want to use it overall, and 
what you already have installed.



We did something similar on a project where we created a simple app 
accessible via AGI, and it stored/retrieved data to/from anXML file.  If 
your access frequency is low enough that might be a good solution.  On 
the other hand if you need complex query capability you should stay on 
the SQL side.



 If you already have MySQL installed for other Asterisk features (eg: 
CDR, or if you use FreePBX) then you might as well use that.


​


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Re: [asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one?

[Антон Сацкий]

Propose U to use Mysql

2015-07-07 17:26 GMT+03:00 Rodrigo Pimenta Carvalho :

>
>
> Hi.
>
> I was studying about how to use databases in Asterisk, accessing it from
> the dial plan.
> In my project, my dial plan will have to store simple data (ex: IP number,
> port number, device name, etc) in a persistent way, so that it will be
> possible to retrieve such information in future moments, still via dial
> plan.
>
> For this case, I would like to know?
>
> 1. What is the best choice for storing and retrieving simple data , with
> dial plan instructions: SQLite or the builtin database option? Consider
> that I'm worried about installation, configuration and use difficulties.
>
> 2. Does Asterisk 13 come with SQLite ready for use or have I to install
> this database separately and configure it to be accessible in dial plan?
>
> 3. Where can I find tutorials about using SQLite or the builtin database
> for storing simple that?
>
> P.S.: I'm not interested in storing CDR data.
>
> Any hint will be very helpful!
>
> Thanks a lot!
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979(Brasil)
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Antony
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моб (063) 656-43-40
satski...@gmail.com 
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[asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one?

[Rodrigo Pimenta Carvalho]

Hi.

I was studying about how to use databases in Asterisk, accessing it from the 
dial plan. 
In my project, my dial plan will have to store simple data (ex: IP number, port 
number, device name, etc) in a persistent way, so that it will be possible to 
retrieve such information in future moments, still via dial plan. 

For this case, I would like to know?

1. What is the best choice for storing and retrieving simple data , with dial 
plan instructions: SQLite or the builtin database option? Consider that I'm 
worried about installation, configuration and use difficulties.

2. Does Asterisk 13 come with SQLite ready for use or have I to install this 
database separately and configure it to be accessible in dial plan?

3. Where can I find tutorials about using SQLite or the builtin database for 
storing simple that?

P.S.: I'm not interested in storing CDR data.

Any hint will be very helpful!

Thanks a lot!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979(Brasil)
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Re: [asterisk-users] How to enable IM over the asterisk server

[Thyda ENG]

Actually, I am using the openfire and I create two users with the SIP
mapping on the openfire to the asterisk server. I can register one user
with the openfire client(Spark) and yes it is connect to asterisk SIP
also.  But with the other one user, I  register it with the SIP
client(Zoiper/ or Linphone) and then I can make the call over these two SIP
but they cannot reach the chat. I wonder what should I config between
openfire and asterisk to enable chat over these two sip clients ?
I am waiting for your reply, Thank.

Thyda

On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland <
kvandenouwel...@vangenechten.com> wrote:

> Good morning Thyda;
>
> Perhaps somebody has a solution for using it on Asterisk itself but after
> some trying I added the Openfire server as a IM server.
>
> I was a bit afraid that 'if' I got it working properly we had to maintain
> it and off course had to troubleshoot it in case it didn't work anymore.
>
> I've read something that you add a ams_msg context in extensions.conf but
> that didn't work for me unfortunaly. It did work for SIP Messages on phones
> but not for IM.
>
> I found Openfire easier to configure and it added a full integration with
> our LDAP which allowed single sign so that users could use the same
> password and log on automatically with the Jitsi client.
>
> But if you have some specific questions, I will be glad to answer.
>
> //Kristof
> >>> Thyda ENG  7/07/2015 6:07 >>>
>  I am currently, I create the VOIP server which enable the user to make
> the call over the asterisk server, Additionally now I want the user to be
> able to chat to each other too.
> I found some suggestion of using the openfire with asterisk but not much
> said on it, Anyway could you please share me how can I config the IM server
> over asterisk?
>
> I am waiting for your reply,
>
> Thyda
>
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> *Cisa Antispam Service*, and is believed to be clean.
>
>
> Privileged Confidential Information may be contained in this message. If
> you are not the addressee indicated in this message (or responsible for
> delivery of the message to such person), you may not copy or deliver this
> message to anyone.
> In such case, you should destroy this message and kindly notify the sender
> by reply email.
> Please advise immediately if you or your employer does not consent to
> Internet email for messages of this kind.
> Opinions, conclusions and other information in this message that do not
> relate to the official business of my firm shall be understood as neither
> given nor endorsed by it.
>
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Re: [asterisk-users] How to enable IM over the asterisk server

[Kristof Van Den Ouweland]

Good morning Thyda;

Perhaps somebody has a solution for using it on Asterisk itself but after some 
trying I added the Openfire server as a IM server.

I was a bit afraid that 'if' I got it working properly we had to maintain it 
and off course had to troubleshoot it in case it didn't work anymore.

I've read something that you add a ams_msg context in extensions.conf but that 
didn't work for me unfortunaly. It did work for SIP Messages on phones but not 
for IM.

I found Openfire easier to configure and it added a full integration with our 
LDAP which allowed single sign so that users could use the same password and 
log on automatically with the Jitsi client.

But if you have some specific questions, I will be glad to answer.

//Kristof
>>> Thyda ENG  7/07/2015 6:07 >>>
I am currently, I create the VOIP server which enable the user to make the call 
over the asterisk server, Additionally now I want the user to be able to chat 
to each other too.
I found some suggestion of using the openfire with asterisk but not much said 
on it, Anyway could you please share me how can I config the IM server over 
asterisk?

I am waiting for your reply,

Thyda

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If you are not the addressee indicated in this message (or responsible for 
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In such case, you should destroy this message and kindly notify the sender by 
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Please advise immediately if you or your employer does not consent to Internet 
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Re: [asterisk-users] Voicemail: saycid without prefix

[A J Stiles]

On Monday 06 Jul 2015, Luca Bertoncello wrote:
> John Kiniston  schrieb:
> > The easiest solution may be to strip the leading zero's off your caller
> > ID before your caller enters the Voicemail app to leave you a message.
> > 
> > 
> > ExecIf(REGEX("^[0][0]."
> > ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))
> 
> Thanks!
> 
> I already had this idea and implemented it.
> It works...

Even better, if the first 4 digits are "0049", you could replace them with "0" 
as though it was an inland call:

ExecIf(REGEX("^0049." 
${CALLERID(NUM)})?Set(CALLERID(num)=0${CALLERID(NUM):4}))

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