Re: [asterisk-users] try to work asterisk 11.11 with ice-upd

[ricky gutierrez]

chan_jingle2 is supported in Asterisk 11?

2014-07-15 13:28 GMT-06:00 ricky gutierrez :
> I'm reading the wiki and says that by default is active, I have it set
> in sip.conf and rtp.conf
>
> icesupport=yes
>
> Usage
>
> By default ICE support is enabled in res_rtp_asterisk. It can be
> explicitly disabled by setting icesupport to no in the rtp.conf
> configuration file.
>
> Icon
>
> ICE support is only used for communication between a remote endpoint
> and Asterisk. It is not used when directmedia is enabled and active
> for a session.
>
> The rtp.conf configuration file also now contains settings for a STUN
> server and TURN server. If these settings are not set support for the
> respective item is disable.
>
> https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support
>
>
>
>
> 2014-07-15 12:41 GMT-06:00 ricky gutierrez :
>> #rpm -qa | grep uuid
>> uuid-1.6.1-10.el6.x86_64
>> libuuid-2.17.2-12.14.el6_5.x86_64
>> uuid-devel-1.6.1-10.el6.x86_64
>>
>> and res_rtp_asterisk was added in the compilation
>>
>> rtp.conf
>>
>> rtpstart=1
>> rtpend=2
>> icesupport=yes
>>
>>
>> 2014-07-15 12:19 GMT-06:00 Joshua Colp :
>>> ricky gutierrez wrote:

 I have configured support for ice in sip.conf, and made a connection
 with motif to jingle, but does not work for me


 [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955
 jingle_interpret_ice_udp_transport: Received ICE-UDP transport
 information on session '8b4hdffbt37vg' but ICE support not available
  -- Executing [s@xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de
 usuario XMPP ") in new stack
>>>
>>>
>>> Do you have the uuid development library installed? It is an optional
>>> dependency and without it res_rtp_asterisk will not be built with ICE
>>> support.
>>>
>>> --
>>> Joshua Colp
>>> Digium, Inc. | Senior Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>> Check us out at: www.digium.com & www.asterisk.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
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> rickygm
>>
>> http://gnuforever.homelinux.com
>
>
>
> --
> rickygm
>
> http://gnuforever.homelinux.com



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Re: [asterisk-users] try to work asterisk 11.11 with ice-upd

[ricky gutierrez]

I'm reading the wiki and says that by default is active, I have it set
in sip.conf and rtp.conf

icesupport=yes

Usage

By default ICE support is enabled in res_rtp_asterisk. It can be
explicitly disabled by setting icesupport to no in the rtp.conf
configuration file.

Icon

ICE support is only used for communication between a remote endpoint
and Asterisk. It is not used when directmedia is enabled and active
for a session.

The rtp.conf configuration file also now contains settings for a STUN
server and TURN server. If these settings are not set support for the
respective item is disable.

https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support




2014-07-15 12:41 GMT-06:00 ricky gutierrez :
> #rpm -qa | grep uuid
> uuid-1.6.1-10.el6.x86_64
> libuuid-2.17.2-12.14.el6_5.x86_64
> uuid-devel-1.6.1-10.el6.x86_64
>
> and res_rtp_asterisk was added in the compilation
>
> rtp.conf
>
> rtpstart=1
> rtpend=2
> icesupport=yes
>
>
> 2014-07-15 12:19 GMT-06:00 Joshua Colp :
>> ricky gutierrez wrote:
>>>
>>> I have configured support for ice in sip.conf, and made a connection
>>> with motif to jingle, but does not work for me
>>>
>>>
>>> [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955
>>> jingle_interpret_ice_udp_transport: Received ICE-UDP transport
>>> information on session '8b4hdffbt37vg' but ICE support not available
>>>  -- Executing [s@xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de
>>> usuario XMPP ") in new stack
>>
>>
>> Do you have the uuid development library installed? It is an optional
>> dependency and without it res_rtp_asterisk will not be built with ICE
>> support.
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.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
>
>
>
> --
> rickygm
>
> http://gnuforever.homelinux.com



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Re: [asterisk-users] try to work asterisk 11.11 with ice-upd

[ricky gutierrez]

#rpm -qa | grep uuid
uuid-1.6.1-10.el6.x86_64
libuuid-2.17.2-12.14.el6_5.x86_64
uuid-devel-1.6.1-10.el6.x86_64

and res_rtp_asterisk was added in the compilation

rtp.conf

rtpstart=1
rtpend=2
icesupport=yes


2014-07-15 12:19 GMT-06:00 Joshua Colp :
> ricky gutierrez wrote:
>>
>> I have configured support for ice in sip.conf, and made a connection
>> with motif to jingle, but does not work for me
>>
>>
>> [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955
>> jingle_interpret_ice_udp_transport: Received ICE-UDP transport
>> information on session '8b4hdffbt37vg' but ICE support not available
>>  -- Executing [s@xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de
>> usuario XMPP ") in new stack
>
>
> Do you have the uuid development library installed? It is an optional
> dependency and without it res_rtp_asterisk will not be built with ICE
> support.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.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



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Re: [asterisk-users] try to work asterisk 11.11 with ice-upd

[Joshua Colp]

ricky gutierrez wrote:

I have configured support for ice in sip.conf, and made a connection
with motif to jingle, but does not work for me


[Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955
jingle_interpret_ice_udp_transport: Received ICE-UDP transport
information on session '8b4hdffbt37vg' but ICE support not available
 -- Executing [s@xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de
usuario XMPP ") in new stack


Do you have the uuid development library installed? It is an optional 
dependency and without it res_rtp_asterisk will not be built with ICE 
support.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] try to work asterisk 11.11 with ice-upd

[ricky gutierrez]

I have configured support for ice in sip.conf, and made a connection
with motif to jingle, but does not work for me


[Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955
jingle_interpret_ice_udp_transport: Received ICE-UDP transport
information on session '8b4hdffbt37vg' but ICE support not available
-- Executing [s@xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de
usuario XMPP ") in new stack

motif.conf
[jingle]
context=xmpp-in
transport=ice-udp
allow=ulaw
allow=alaw
allow=h263
allow=h264
connection=admin

any idea?



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Re: [asterisk-users] Call drop on Aastra SIP phones

[Bruno Rocha]

Hi Joshua!

On 2014-07-15 13:39, Joshua Colp wrote:

Bruno Rocha wrote:

Hello everybody,


Hola,


I'm having issues with calls being dropped on Aastra phones, when the
call is on hold. Tested with models 6863i and 6867i.
I've figured that the call is dropped by Asterisk when it reaches the
rtpholdtimeout limit.

I've reported the issue to Aastra, asking them to implement some kind of
"RTP keep-alive" feature on their phones. Maybe the phone could send
some RTCP frame (or an empty RTP frame) just to prove it is alive.
Unfortunately Aastra said the hold behaviour on the phone is correct, as
per RFC 3264, section 8.4, 4th paragraph:

Typically, when a user "presses" hold, the agent will generate an
offer with all streams in the SDP indicating a direction of sendonly,
and it will also locally mute, so that no media is sent to the far
end, and no media is played out.


They are correct. The "rtpholdtimeout" option stems from a time when it
was not possible to monitor the signaling of the call and is an
Asterisk-ism. You've got a few options, though:

1. Increase the rtpholdtimeout
2. Don't use rtpholdtimeout and use SIP session timers instead (check
the SIP Session-Timers section in sip.conf.sample)



Thanks for the clarification! I will try the SIP Session-Timers.

Cheers,
--
Bruno Rocha

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[asterisk-users] Queue wrapuptime and active calls

[Eduardo Leones]

Hello,

My question is this, I have a service queue that members follow the service
interval (wrapuptime = 30).

However, sometimes these members need to call the customer back, thus
making an active call. Occurs when this member disconnects the call shortly
following section in the queue already sends a new call for him not
understanding the range (wrapuptime = 30).

My question is if I can somehow make the queue respect the same range with
a call that was not caused by the queue. Is there any way I wrapuptime in a
reset after a manual call?

tks

Eduardo
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Re: [asterisk-users] Need to spoof the callerid using the AMI Originate

[Dan Cropp]

Thank you Tiago.

I was been able to set the callerid name and number using the Originate 
action’s variables.

The problem I am currently running into is the SIP messages RPID portion always 
indicates the caller id has not been screened.  Even when I set the 
CALLERID(name-pres) (or CALLERID(num-pres) variables to 
allowed_passed_screened, the RPID portion of the SIP message always indicates 
it has not passed screening.

The only way I can get that setting to be passed to the SIP Provider the way 
our Freeswitch does is if I do as you describe (additionally setting the 
name-pres/num-pres variables in the local dialplan).
It would be convenient if the Originate call had the ability to toggle those 
flags.  At least there is a solution.

Have a great day!
Dan

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Sent: Sunday, July 06, 2014 1:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need to spoof the callerid using the AMI Originate

Actually you shold do that on "MyContext"

On 6 July 2014 19:21, Tiago Geada 
mailto:tiago.ge...@gmail.com>> wrote:
Hi,

You can use a Local channel in your  originate, and have a piece of local 
dialplan change that for you. Set(CALLERID(num)=x)

On 13 June 2014 15:32, Dan Cropp mailto:d...@amtelco.com>> 
wrote:
We have several customers we need to place outbound calls for (in a single 
system).  May have to place calls for thousands of different caller ids.  
Customer signs a contract guaranteeing the caller id they provide is owned by 
them.

I have everything setup for AMI Originate and can place the calls.

However, I’m encountering a problem with the caller id.
The system I’m dialing through requires my contact to be something like 
1234@xyz.

If I set the CallerID to something like Jane Doe <1234> it will correctly set 
my Contact so the system accepts the call and it dials the number.
However, the SIP INVITE message From field is set to “Jane Doe” 
1...@xxx.xxx.xxx.xxx
Is there a way to make the SIP INVITE message have different caller id values 
for the From and the Contact fields?

Additionally, is it possible to set the callerid number value to a PSTN number 
instead of a SIP number@domain?

I tried setting the callerid(num) via the variable field, but that doesn’t seem 
to work.
Below is a sample of the AMI Originate message I’m sending.

Action: Originate
ActionID: MyAction
Channel: SIP/xxx.xxx.xxx.xxx/1234567890
Exten: testing
Context: MyContext
Priority: 1
Timeout: 3
CallerID: Jane Doe <123>
Variable: CALLERID(num)=222333
Async: true


Have a great day!
Dan

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Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

[Olli Heiskanen]

Wow, thanks Joshua, it would've taken me forever to find the answer there.
It did the trick and the registrations look much better.

Merci beaucoup!

- Olli



2014-07-15 16:26 GMT+03:00 Joshua Colp :

> Olli Heiskanen wrote:
>
>>
>> Thanks, there are no register lines in my sip.conf, but I have defined
>> callbackextension fields in the realtime table, to be the same value as
>> the extension name. In this case, extension 771 has callbackextension
>> value 771. I tried replacing those with null values but that had no
>> effect on the outcome.
>>
>
> The callbackextension is the reason this is happening.
>
> From sip.conf.sample:
>
> ; A similar effect can be achieved by adding a "callbackextension" option
> in a peer section.
> ; this is equivalent to having the following line in the general section:
> ;
> ;register => username:secret@host/callbackextension
> ;
> ; and more readable because you don't have to write the parameters in two
> places
> ; (note that the "port" is ignored - this is a bug that should be fixed).
>
> Remove that column from your table, restart Asterisk, and it should go
> away.
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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> _
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Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

[Joshua Colp]

Olli Heiskanen wrote:


Thanks, there are no register lines in my sip.conf, but I have defined
callbackextension fields in the realtime table, to be the same value as
the extension name. In this case, extension 771 has callbackextension
value 771. I tried replacing those with null values but that had no
effect on the outcome.


The callbackextension is the reason this is happening.

From sip.conf.sample:

; A similar effect can be achieved by adding a "callbackextension" 
option in a peer section.

; this is equivalent to having the following line in the general section:
;
;register => username:secret@host/callbackextension
;
; and more readable because you don't have to write the parameters in 
two places

; (note that the "port" is ignored - this is a bug that should be fixed).

Remove that column from your table, restart Asterisk, and it should go away.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

[Olli Heiskanen]

Thanks, there are no register lines in my sip.conf, but I have defined
callbackextension fields in the realtime table, to be the same value as the
extension name. In this case, extension 771 has callbackextension value
771. I tried replacing those with null values but that had no effect on the
outcome.

Currently when I register clients in, after some seconds Asterisk starts
sending REGISTER messages, at which point Kamailio sees 2 AORs, here's an
example:
(here 1.1.1.1 is the public ip of my server that houses Kamailio at 5060
and Asterisk at 5070, and 2.2.2.2 is the public ip of the network clients
are in)

AOR:: 7...@testers.com
Contact::
sip:771@2.2.2.2:5060;rinstance=c8447637c890c010;transport=UDP
Q=
Expires:: 3470
Callid::
NDQ5Njk4ZmUxZGJhNzRjMzUwMTA2OThmOGFjYzc4Zjk.
Cseq:: 2
User-agent:: Z 3.2.21357 r21367
State:: CS_SYNC
Flags:: 0
Cflag:: 0
Socket:: udp:1.1.1.1:5060
Methods:: 5087
Ruid:: uloc-53bfe447-35b0-608
Reg-Id:: 0
Last-Keepalive:: 1405429865
Last-Modified:: 1405429865
AOR:: 771@1.1.1.1
Contact:: sip:771@1.1.1.1:5070 Q=
Expires:: 105
Callid::
3e946958322b1e2d6bfa564d46bf8...@testers.com
Cseq:: 133
User-agent:: Asterisk PBX
State:: CS_SYNC
Flags:: 0
Cflag:: 0
Socket:: udp:1.1.1.1:5060
Methods:: 4294967295
Ruid:: uloc-53bfe447-35b0-708
Reg-Id:: 0
Last-Keepalive:: 1405429980
Last-Modified:: 1405429980

I guess there should be only one AOR, so Asterisk might get wrong kind of
data to begin with or it's configured incorrectly. In my sip trace the
REGISTER flow from client to Kamailio to Asterisk seems ok, I could be
wrong though.

In my setup clients authenticate with Kamailio and Kamailio sends a
REGISTER to Asterisk according to guide I used:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

How would I fix this double-AOR problem, can it be fixed on Asterisk
configuration?

thanks,
Olli







2014-07-15 16:00 GMT+03:00 Joshua Colp :

> Olli Heiskanen wrote:
>
>> Hello,
>>
>> Thanks for your response, I actually verified that the Zoiper setting is
>> not the reason for Asterisk to start sending REGISTERs, it only looked
>> like it as I checked the Kamailio output before Asterisk sent the first
>> REGISTER to Kamailio, right after I had played with that setting.
>> (sorry, my bad!)
>>
>> However, _something_ is causing these REGISTERs, here's an example of a
>> REGISTER message sent from Asterisk to Kamailio:
>>
>> REGISTER sip:testers.com  SIP/2.0
>>
>>  Via: SIP/2.0/UDP my_ip:5070;branch=z9hG4bK7477f754;rport
>>  Max-Forwards: 70
>>  From: ;tag=as7a88c4c6
>>  To: 
>>  Call-ID: 3e946958322b1e2d6bfa564d46bf8...@testers.com
>> 
>>
>>  CSeq: 121 REGISTER
>>  User-Agent: Asterisk PBX
>>  Expires: 120
>>  Contact: > >
>>
>>  Content-Length: 0
>>
>> Is there any other reason - other than client settings - why this would
>> happen?
>>
>
> If Asterisk was configured to do so, yes. Do you have any register lines
> in sip.conf or do you have the "callbackextension" option set for any peers?
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] Call didn't stop after losing one leg

[lm]

Hi Joshua, 

> Since you are having media go directly the only thing that can be
> monitored is the signaling of the call itself. This can be
> accomplished using SIP session timers. There is a section "SIP
> Session-Timers" in the sip.conf.sample file which has the various
> configuration options relating to it.

I have been reading about it and it looks like an effective way to 
mitigate this problem. I'll test it and see how it goes with my providers.

Thank you!!

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Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

[Joshua Colp]

Olli Heiskanen wrote:

Hello,

Thanks for your response, I actually verified that the Zoiper setting is
not the reason for Asterisk to start sending REGISTERs, it only looked
like it as I checked the Kamailio output before Asterisk sent the first
REGISTER to Kamailio, right after I had played with that setting.
(sorry, my bad!)

However, _something_ is causing these REGISTERs, here's an example of a
REGISTER message sent from Asterisk to Kamailio:

REGISTER sip:testers.com  SIP/2.0
 Via: SIP/2.0/UDP my_ip:5070;branch=z9hG4bK7477f754;rport
 Max-Forwards: 70
 From: ;tag=as7a88c4c6
 To: 
 Call-ID: 3e946958322b1e2d6bfa564d46bf8...@testers.com

 CSeq: 121 REGISTER
 User-Agent: Asterisk PBX
 Expires: 120
 Contact: http://sip:771@91.221.66.61:5070>>
 Content-Length: 0

Is there any other reason - other than client settings - why this would
happen?


If Asterisk was configured to do so, yes. Do you have any register lines 
in sip.conf or do you have the "callbackextension" option set for any peers?


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Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

[Olli Heiskanen]

Hello,

Thanks for your response, I actually verified that the Zoiper setting is
not the reason for Asterisk to start sending REGISTERs, it only looked like
it as I checked the Kamailio output before Asterisk sent the first REGISTER
to Kamailio, right after I had played with that setting. (sorry, my bad!)

However, _something_ is causing these REGISTERs, here's an example of a
REGISTER message sent from Asterisk to Kamailio:

REGISTER sip:testers.com SIP/2.0
Via: SIP/2.0/UDP my_ip:5070;branch=z9hG4bK7477f754;rport
Max-Forwards: 70
From: ;tag=as7a88c4c6
To: 
Call-ID: 3e946958322b1e2d6bfa564d46bf8...@testers.com
CSeq: 121 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: 
Content-Length: 0

Is there any other reason - other than client settings - why this would
happen?

cheers,
Olli



2014-07-15 15:40 GMT+03:00 Joshua Colp :

> Olli Heiskanen wrote:
>
>>
>> Hello all,
>>
>
> Bonjour,
>
>
>  I have an Asterisk installation with Kamailio using realtime
>> integration. I have gotten my clients to register, but there is
>> something odd about the sip message flow with some of my clients. My
>> clients are Zoiper and Asterisk is 11.10.2.
>>
>> When I set 'Subscribe to MWI' value to 'both', after a normal,
>> successful registration Asterisk begins to send REGISTER messages to
>> Kamailio every 105 seconds. Kamailio responds with 200 OK. If I set the
>> value to 'disabled', Asterisk does not send these frequent REGISTER
>> messages. Probably due to these REGISTERs Kamailio sees 2 AORs for the
>> account for those clients whose 'Subscribe to MWI' setting is defined as
>> 'both'.
>>
>
> Can you provide a link to a sip debug log of this occurring? It sounds
> extremely weird and I'm not really sure how chan_sip would be doing such a
> thing...
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

[Joshua Colp]

Olli Heiskanen wrote:


Hello all,


Bonjour,


I have an Asterisk installation with Kamailio using realtime
integration. I have gotten my clients to register, but there is
something odd about the sip message flow with some of my clients. My
clients are Zoiper and Asterisk is 11.10.2.

When I set 'Subscribe to MWI' value to 'both', after a normal,
successful registration Asterisk begins to send REGISTER messages to
Kamailio every 105 seconds. Kamailio responds with 200 OK. If I set the
value to 'disabled', Asterisk does not send these frequent REGISTER
messages. Probably due to these REGISTERs Kamailio sees 2 AORs for the
account for those clients whose 'Subscribe to MWI' setting is defined as
'both'.


Can you provide a link to a sip debug log of this occurring? It sounds 
extremely weird and I'm not really sure how chan_sip would be doing such 
a thing...


Cheers,

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Call drop on Aastra SIP phones

[Joshua Colp]

Bruno Rocha wrote:

Hello everybody,


Hola,


I'm having issues with calls being dropped on Aastra phones, when the
call is on hold. Tested with models 6863i and 6867i.
I've figured that the call is dropped by Asterisk when it reaches the
rtpholdtimeout limit.

I've reported the issue to Aastra, asking them to implement some kind of
"RTP keep-alive" feature on their phones. Maybe the phone could send
some RTCP frame (or an empty RTP frame) just to prove it is alive.
Unfortunately Aastra said the hold behaviour on the phone is correct, as
per RFC 3264, section 8.4, 4th paragraph:

Typically, when a user "presses" hold, the agent will generate an
offer with all streams in the SDP indicating a direction of sendonly,
and it will also locally mute, so that no media is sent to the far
end, and no media is played out.


They are correct. The "rtpholdtimeout" option stems from a time when it 
was not possible to monitor the signaling of the call and is an 
Asterisk-ism. You've got a few options, though:


1. Increase the rtpholdtimeout
2. Don't use rtpholdtimeout and use SIP session timers instead (check 
the SIP Session-Timers section in sip.conf.sample)


Cheers,

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Re: [asterisk-users] Call didn't stop after losing one leg

[Joshua Colp]

lm wrote:

Hello there,


Kia ora,


I'm using a Debian box with Asterisk 1.8.13.1 as a DID-PSTN gateway,
so I can receive calls in a DID number and redirect it to my mobile line.

It has been working flawlessly for a few months, but I have noticed
that some calls were not cut after losing one leg (the one with the
DID server), and kept the PSTN leg active until I restarted the
server (with the unexpected cost involved in the PSTN call).

The relevant extensions.conf line is:

exten =>  34911234567,1,Dial(SIP/pstn/447123456789)

And both DID and PSTN sip accounts have canreinvite=yes, so they
can have direct media.

I haven't collected any debug log nor any other relevant information.

Does anybody know why something like this happens, or how can I
cut a call that unexpectedly losed a leg?


Since you are having media go directly the only thing that can be 
monitored is the signaling of the call itself. This can be accomplished 
using SIP session timers. There is a section "SIP Session-Timers" in the 
sip.conf.sample file which has the various configuration options 
relating to it.


Cheers,

--
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Recording sound.

[Joshua Colp]

Anurag Rana wrote:

Hi All,


Kia ora,


I am calling mobile numbers from Soft-phone and recording the call.
In recording the level of sound from the receiver's side is perfect
(loud enough) but my voice's sound level is very weak. I barely can hear
it.

During the call receiver is able to hear me. But in recording my part of
conversation is barely audible.

I am recording using MixMonitor().


Asterisk does not modify the volume of the audio itself in MixMonitor. 
It's likely that the audio is coming in low already.



Is there anything that can be done to mitigate the problem?


You can try using the VOLUME dialplan function to increase the volume 
manually, but this will impact what you hear as well.


Cheers,

--
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[asterisk-users] Corrupted sqlite3 astdb back end

[Mauricio Tavares]

  So I found out my astdb database is boink (asterisk 1.8.something):

[Jul 15 06:42:28] ERROR[18769] res_config_sqlite.c: database disk
image is malformed

# first stop asterisk before doing this, and do it on a copy:

sqlite3 ./sqlite.db .dump |less
PRAGMA foreign_keys=OFF;
BEGIN TRANSACTION;
/ ERROR: (26) file is encrypted or is not a database */
ROLLBACK; -- due to errors
(END)

Anything I can do to repair it? If not, how can I recreate it? What is
stored in it (https://wiki.asterisk.org/wiki/display/AST/SQLite3+astdb+back-end
was not particularly helpful)?

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[asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

[Olli Heiskanen]

Hello all,

I have an Asterisk installation with Kamailio using realtime integration. I
have gotten my clients to register, but there is something odd about the
sip message flow with some of my clients. My clients are Zoiper and
Asterisk is 11.10.2.

When I set 'Subscribe to MWI' value to 'both', after a normal, successful
registration Asterisk begins to send REGISTER messages to Kamailio every
105 seconds. Kamailio responds with 200 OK. If I set the value to
'disabled', Asterisk does not send these frequent REGISTER messages.
Probably due to these REGISTERs Kamailio sees 2 AORs for the account for
those clients whose 'Subscribe to MWI' setting is defined as 'both'.

I know this is mostly not an Asterisk problem, but I'd like to understand
better what exactly makes Asterisk to react this way? I didn't find any
differences in the SIP messages during registration (I could be just blind
though...), or in the way the clients are set up in the Realtime db.

On the 'Subscribe to MWI' setting the Zoiper documentation states: "this
tag specifies when Zoiper is going to subscribe for Message Waiting
Indication(MWI) for this account". In addition to values 'both' and
'disabled' there are values 'before registration (Asterisk)' and 'after
registration'. To me it seems strange to use REGISTER messages for
subscribing to something related to voicemail messages. So far I haven't
learned about the way Asterisk handles voicemail stuff but if You guys have
some clarification on why I'm getting these results I'd appreciate it!

cheers,
Olli
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[asterisk-users] Which GUI to query CDR when stored in radius ?

[Olivier]

Hello,

I've seen a couple a VoIP equipment (Patton) which store CDR data into
radius database.
As Asterisk is also capable of doing so, I'm wondering if it's worth
standardizing on it.

1. How can you query or build custom reports with CDR data in radius
database ?
Which tools do you use for that ?
I've seen some software like Daloradius "dealing with radius data" but,
from a quick glance, I've not seen anything specific for telephony.

2. Given radius expendability, shall I expect major differences between CDR
produced by equipment from different vendors ?

Regards
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Re: [asterisk-users] http://downloads.asterisk.org/pub/telephony/dahdi-linux/ not showing lastest changelog [SOLVED]

[Olivier]

Thanks for replying.


2014-07-14 17:54 GMT+02:00 Russ Meyerriecks :

> On Mon, Jul 14, 2014 at 2:56 AM, Olivier  wrote:
> > Is it me or unlike http://downloads.asterisk.org/pub/telephony/asterisk/
> ,
> > http://downloads.asterisk.org/pub/telephony/dahdi-linux/ is not showing
> > lastest changelog anymore ?
>
> dahdi project dropped generating this specific changelog during the
> svn to git migration. Honestly, we didn't think anybody used it and
> the information is replicated elsewhere.
>
> diffstats and patch shortlogs are generated for each release email. We
> also keep sane shortlog synopsis for each patch which makes it a lot
> easier to browse the git logs here, if you're looking for something
> specific:
>
> http://git.asterisk.org/gitweb/?p=dahdi/linux.git
> http://git.asterisk.org/gitweb/?p=dahdi/tools.git
>
>
> --
> Russ Meyerriecks
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> direct: +1 256-428-6025
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>
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