Re: [asterisk-users] PJSIP is Ignored

2016-08-12 Thread George Joseph 
On Fri, Aug 12, 2016 at 12:53 PM, George Joseph  wrote:

>
> On Fri, Aug 12, 2016 at 12:02 PM, Saint Michael  wrote:
>
>> ​Asterisk 13.11 rc1
>>
>> ./configure   LDFLAGS="-z muldefs" --libdir=/usr/lib64
>> --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode
>> --with-pjproject-bundled
>>
>>
Oddly enough, it's --disable-dev-mode.   Try leaving that out and see what
happens.  The two shouldn't be related at all.



> ​checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no
>> checking for pjsip_tsx_create_uac2 in -lpjsip... no
>> checking if "pjmedia_mod_offer_flag flag = 
>> PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE"
>> compiles using pjmedia.h... no
>> checking for pjsip_get_dest_info in -lpjsip... no
>> checking for pjsip/include/pjsip/sip_util.h in -lpj... no
>> checking for pjsip_endpt_set_ext_resolver in -lpjsip... no
>> checking if "struct pjsip_tls_setting setting; int proto; proto =
>> setting.proto;" compiles using pjsip.h... no
>> checking for pjsip_evsub_add_ref in -lpjsip... no
>>
>> ​When I do make menuselect, it is disabled.
>>
>
> Well, that's bizarre.  I get the same result.
> Would you open an issue at issues.asterisk.org while I look into it?
>
>
>
>
>> ​
>>
>>
>> --
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>
>
>
> --
> George Joseph
> Digium, Inc. | 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] PJSIP is Ignored

2016-08-12 Thread George Joseph 
On Fri, Aug 12, 2016 at 12:02 PM, Saint Michael  wrote:

> ​Asterisk 13.11 rc1
>
> ./configure   LDFLAGS="-z muldefs" --libdir=/usr/lib64
> --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode
> --with-pjproject-bundled
>
> ​checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no
> checking for pjsip_tsx_create_uac2 in -lpjsip... no
> checking if "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE"
> compiles using pjmedia.h... no
> checking for pjsip_get_dest_info in -lpjsip... no
> checking for pjsip/include/pjsip/sip_util.h in -lpj... no
> checking for pjsip_endpt_set_ext_resolver in -lpjsip... no
> checking if "struct pjsip_tls_setting setting; int proto; proto =
> setting.proto;" compiles using pjsip.h... no
> checking for pjsip_evsub_add_ref in -lpjsip... no
>
> ​When I do make menuselect, it is disabled.
>

Well, that's bizarre.  I get the same result.
Would you open an issue at issues.asterisk.org while I look into it?




> ​
>
>
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[asterisk-users] PJSIP is Ignored

2016-08-12 Thread Saint Michael 
​Asterisk 13.11 rc1

./configure   LDFLAGS="-z muldefs" --libdir=/usr/lib64
--with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode
--with-pjproject-bundled

​checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no
checking for pjsip_tsx_create_uac2 in -lpjsip... no
checking if "pjmedia_mod_offer_flag flag =
PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using pjmedia.h... no
checking for pjsip_get_dest_info in -lpjsip... no
checking for pjsip/include/pjsip/sip_util.h in -lpj... no
checking for pjsip_endpt_set_ext_resolver in -lpjsip... no
checking if "struct pjsip_tls_setting setting; int proto; proto =
setting.proto;" compiles using pjsip.h... no
checking for pjsip_evsub_add_ref in -lpjsip... no

​When I do make menuselect, it is disabled.
​
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Re: [asterisk-users] losing audio from one end after 5 min.

2016-08-12 Thread D'Arcy J.M. Cain 
On Fri, 12 Aug 2016 16:44:32 +
"Jonas Christoffersen"  wrote:
> Just tested the connection in the other direction and when calling
> out there is no problem.
> only when calling in.

It sure sounds like a NAT problem.  Missing audio has a 90% or more
probability of being NAT related.  Maybe it's a problem on the gateway
device.  What is the modem/router?

P.S. The subject was driving me nuts.  I had to correct it.

-- 
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http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens 

Hello


running into several problems when installing 
asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 12).


I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled

First, I do not seem to have res_srtp module available, although all 
necessary libs are present on the system


Second, I am not able to start Asterisk with following error : 
"/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2: 
cannot open shared object file: No such file or directory"





Help appreciated.

Kind regards.




On 12-08-16 16:58, Jonas Kellens wrote:


On 12-08-16 16:38, Joshua Colp wrote:

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm 
having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. 
It's okay if pjproject doesn't as we don't use their media layer. Do 
you have the res_srtp module in Asterisk?




Hello

Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest 
version

Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version

However, I am not able to select res_srtp module in menuselect. It 
says XXX res_srtp module




Kind regards.





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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens 


On 12-08-16 16:38, Joshua Colp wrote:

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm 
having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. 
It's okay if pjproject doesn't as we don't use their media layer. Do 
you have the res_srtp module in Asterisk?




Hello

Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest 
version

Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version

However, I am not able to select res_srtp module in menuselect. It says 
XXX res_srtp module




Kind regards.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Joshua Colp 

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. It's 
okay if pjproject doesn't as we don't use their media layer. Do you have 
the res_srtp module in Asterisk?


--
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] Switching between Music on Hold streams. [13.8.2]

2016-08-12 Thread Jonathan H 
Hello!

I thought having finally "cracked it", I might as well post what I've done.

https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/moh-switching.md

Can someone please take a quick look and see if there's anything I could
have done better or more efficiently, or if anything stands out as
particularly horrific?

Basically, it uses an app called crudini to add sections to
musiconhold.conf, then does an moh reload.

When the user has finished listening and presses * then the remote
extension is dropped and the caller returns to the current menu.

The nice thing about this is that even if two callers call and listen to
the same moh stream, when one hangs up, even though it deletes the config
and reloads moh, Asterisk is nice to the other caller and they keep
listening.

The end result is what I wanted, which is to not have any extra CPU load or
network usage when no-one is listening. And if more than one person is
listening, it's still only "using" one remote stream, as I've uncommented
the cachertclasses value.


[general]
cachertclasses=yes ; use 1 instance of moh class for all users who are
using it


As a little bonus, I've put what I think is a clever little "menu maker"
in, which grabs and caches short audio files using the free plan from
voicerss.org.

If anyone wants to try it in practice, call UK +44 20 36 37 60 70 - this
number is working as of the 12th of August and I'll leave it up at least
over the weekend, but if you're reading this in a few weeks, don't expect
it to work! (I'm allowed to use these streams before anyone panics!).

On 11 May 2016 at 11:09, Dovid Bender  wrote:

> If you ever figure out AAC in Asterisk for MOH let me know. The ones that
> I have working is MP3 and MMS.
>
> On Mon, May 9, 2016 at 1:18 PM, Jonathan H  wrote:
>
>> Thanks Joshua and everyone,
>>
>> Joshua's solution seems a lot simpler and works well. Only one thing
>> now - The reason I named the classes as I did, was so that I could
>> select the class based on callerID plus extension.
>>
>> Unless I've misread it, I'm limited to 9 switchable classes via the
>> "digit=#" option, is that correct?
>>
>> Or is there a clever hack around this?
>>
>> extensions.conf
>>
>> [streamdemo]
>> exten => s,1,Answer
>> exten => s,2,BackGround(menu)
>> exten => s,3,WaitExten
>> exten => _[2,3,4,5],1,MusicOnHold(${CALLERID(name)}${EXTEN})
>> ;exten => s,5,Goto(s,2)
>> exten => _[X,t,i],1,Goto(streamdemo,s,2)
>>
>> and in musiconhold.conf (4 is commented out as it's AAC and I've not
>> figured that one out yet - bonus points to someone who can point the
>> way!)
>>
>> [streamdemo2]
>> mode=custom
>> digit=2
>> application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
>> http://185.14.85.162:8020
>>
>> [streamdemo3]
>> mode=custom
>> digit=3
>> application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
>> http://stream.acbradio.org:8000/mainstream.mp3
>>
>> ;[streamdemo4]
>> ;mode=custom
>> ;digit=4
>> ;application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
>> http://199.180.75.27:80/
>> ;http://www.mushroomfm.com/media/listen.pls
>>
>> [streamdemo5]
>> digit=5
>> mode=custom
>> application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
>> http://206.225.87.121:8000/
>>
>> On 9 May 2016 at 18:00, A J Stiles  wrote:
>> > On Monday 09 May 2016, Jonathan H wrote:
>> >> . {stuff deleted} .
>> >> [streamdemo]
>> >> exten => s,1,Answer
>> >> exten => s,2,BackGround(menu)
>> >> exten => s,3,WaitExten
>> >> exten => s,4,Goto(s,2)
>> >> exten =>
>> >> _[2,3,4,5],1,Dial(Local/${EXTEN}@play-radio,,G(play-radio^${EXTEN}^2))
>> >> exten => _[2,3,4,5],2,Goto(s,2)
>> >
>> > You have an error in your dialplan!  The pattern _[2,3,4,5] will match
>> any of
>> > 2, a comma, 3, a comma  (again), 4, a comma or 5.
>> >
>> > I think you might mean  _[2345]  which will match any of 2, 3, 4 or 5
>> (but
>> > not a comma),  and contains no tautologies.
>> >
>> >
>> > --
>> > AJS
>> >
>> > Note:  Originating address only accepts e-mail from list!  If replying
>> off-
>> > list, change address to asterisk1list at earthshod dot co dot uk .
>> >
>> > --
>> > _
>> > -- 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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>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>
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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens 
Question : I noticed I received an error when installing pjproject 
--with-external-srtp


I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm having ??



Kind regards.


On 12-08-16 15:02, Jonas Kellens wrote:

Hello


setting "nat=no" or omitting "nat=" in peer definition does not help 
either. Still no audio.


Why do you think this is a NAT issue ? IP and port information in 
SDP-body is correct.





Kind regards.


On 12-08-16 09:25, Антон Сацкий wrote:


Try delete nat from 77wrtc settings ice should do the same


On Aug 11, 2016 10:00 PM, "Jonas Kellens" > wrote:


On 11-08-16 18:03, Matt Fredrickson wrote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
>
wrote:

My main reason not to upgrade to Ast 13 is because I'm
afraid of losing
functionality as there are certain functions
deprecated/replaced. This can
also cause headache :-)

I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in
this case ? If ICE
and NAT is working (not causing problems) why should Ast
13 bring me audio
and Ast 12 don't ??

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature
standard
like ISDN or SIP.  If you find interop problems in an older
version of
Asterisk with WebRTC, it's likely that it has been fixed in
13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking
to is
not on the same network as the Asterisk server, it's
*possible* you
might need a true TURN server as well, instead of just an ICE
server.

Matthew Fredrickson


Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp

turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA


then Asterisk 12 gets really slow and sometimes unresponsive.
Calls result in 480 request timeout (possibly due to the freeze
of Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.


(putting the above lines back in comment resolves the issue of
Asterisk freeze. This is all EXTREMELY BUGGY !)


Asterisk 13 here I come (with very high expectations).


Kind regards.


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Re: [asterisk-users] Asterisk & Vitelity Invite issues

2016-08-12 Thread Israel Gottlieb 
Could you please write the problem your having and not the reason to the
problem
Maybe the reason is something else

בתאריך 8 באוג׳ 2016 17:25,‏ "Tammy Firefly"  כתב:

Hi All,

We have asterisk 11.23 running sip to vitelity and from there IAX trunks
split off to where they need to go.  We are having a problem getting
chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
sending a reinvite which their side & they do not support us sending a
reinvite.  Ive tried:

canreinvite=no which was supposedly replaced by:

directmedia=no

Can anyone shed any light on this matter?  I'd love to get this fixed.

There is no firewall on this machine at all.

Thanks
--Tammy

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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens 

Hello


setting "nat=no" or omitting "nat=" in peer definition does not help 
either. Still no audio.


Why do you think this is a NAT issue ? IP and port information in 
SDP-body is correct.





Kind regards.


On 12-08-16 09:25, Антон Сацкий wrote:


Try delete nat from 77wrtc settings ice should do the same


On Aug 11, 2016 10:00 PM, "Jonas Kellens" > wrote:


On 11-08-16 18:03, Matt Fredrickson wrote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
>
wrote:

My main reason not to upgrade to Ast 13 is because I'm
afraid of losing
functionality as there are certain functions
deprecated/replaced. This can
also cause headache :-)

I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in
this case ? If ICE
and NAT is working (not causing problems) why should Ast
13 bring me audio
and Ast 12 don't ??

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature
standard
like ISDN or SIP.  If you find interop problems in an older
version of
Asterisk with WebRTC, it's likely that it has been fixed in
13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking
to is
not on the same network as the Asterisk server, it's
*possible* you
might need a true TURN server as well, instead of just an ICE
server.

Matthew Fredrickson


Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp

turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA


then Asterisk 12 gets really slow and sometimes unresponsive.
Calls result in 480 request timeout (possibly due to the freeze of
Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.


(putting the above lines back in comment resolves the issue of
Asterisk freeze. This is all EXTREMELY BUGGY !)


Asterisk 13 here I come (with very high expectations).


Kind regards.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Антон Сацкий 
Try delete nat from 77wrtc settings ice should do the same

On Aug 11, 2016 10:00 PM, "Jonas Kellens"  wrote:

> On 11-08-16 18:03, Matt Fredrickson wrote:
>
>> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens 
>> wrote:
>>
>>> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
>>> functionality as there are certain functions deprecated/replaced. This
>>> can
>>> also cause headache :-)
>>>
>>> I will do so if there is no other option.
>>>
>>> But still, I don't see why Ast 13 would differ so much in this case ? If
>>> ICE
>>> and NAT is working (not causing problems) why should Ast 13 bring me
>>> audio
>>> and Ast 12 don't ??
>>>
>> If you want to minimize grief, start with 13 - WebRTC has been a
>> moving target for the last 5 years, it is not an old, mature standard
>> like ISDN or SIP.  If you find interop problems in an older version of
>> Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
>> it hasn't the most likely place to obtain the fix will be in 13.
>>
>> After you get the WebRTC part working, then you can move back the
>> versions of Asterisk you're using to see if it still works.
>>
>> As far as ICE not working goes, if the browser you're talking to is
>> not on the same network as the Asterisk server, it's *possible* you
>> might need a true TURN server as well, instead of just an ICE server.
>>
>> Matthew Fredrickson
>>
>>
> Matthew
>
> when I set the following in rtp.conf :
>
> turnaddr=192.158.29.39:3478?transport=udp
> turnusername=28224511:1379330808
> turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA
>
>
> then Asterisk 12 gets really slow and sometimes unresponsive. Calls result
> in 480 request timeout (possibly due to the freeze of Asterisk).
>
> So this is also no solution.
>
> Can not even test if it brings me some audio in my webRTC calls.
>
>
> (putting the above lines back in comment resolves the issue of Asterisk
> freeze. This is all EXTREMELY BUGGY !)
>
>
> Asterisk 13 here I come (with very high expectations).
>
>
> Kind regards.
>
>
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