Re: [asterisk-users] Asterisk 13 and WebRTC

[Yuriy Gorlichenko]

Hi. It have big audio delay because using extenral ICE servers.
Better to use kamailio/opensips + rpenigne infront

2016-09-09 0:36 GMT+03:00 Annus Fictus :

> Hello list,
>
> before to lost my time, I'd like know if someone have a WebRTC working
> configuration on Asterisk 13.11.0 SIP or PJSIP channel.
>
> Thank you
>
> Regards
>
>
>
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[asterisk-users] Sending bye to not establishment session

[Yuriy Gorlichenko]

Hello.We have an issue with canseling dialogs.

Scenario that we have issue is:

Calling to some extensions from endpoint

Hanging Up caller party until ringing send to asterisk from second leg
(called party)
Asterisk resend Bye to called party but Bye not going to somewhere because
no Sip dialog from called party stated.

I think asterisk should send Cancel to called party when this happends.
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Re: [asterisk-users] MixMonitor with b option recording all calls

[Yuriy Gorlichenko]

SIP trunks

2014-09-22 12:12 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk:

 On Monday 22 Sep 2014, Yuriy Gorlichenko wrote:
  Hello I have an issue wit MixMonitor. I need to record only answered
 calls,
  so I set b option for this but calls still recording even call no
  answered My asterisk version 12.5.1, at my other servers with older
  versions of asterisk (11.8 for example) MixMonitor works fine.

 What technology are you using for your outgoing calls?  SIP trunk, IAX
 trunk,
 ISDN, mobile or analogue phone lines?

 --
 AJS

 Note:  Originating address only accepts e-mail from list!  If replying off-
 list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] MixMonitor with b option recording all calls

[Yuriy Gorlichenko]

my sip.conf
[kamailio_ext1]
type=friend
host = my.superprovider.com
port = 5068
canreinvite = no
insecure = invite,port
transport=udp
trustrpid=yes
context = incoming
videosupport=no
directmedia=no
dtlsenable = no
tlsenable=no
disallow=all
allow=alaw
allow=opus
allow=ulaw

connection goes great. SIP session have all packets

So astersik send INVITE - to trunk
then
- TRYING
- RINGING
- OK
-ACK

and at the end some of callers send BYE

And then Goes OK and ACK

full SIP session. Signaling is Ok. At CDR I see ANSWERED

Wheb Call Unanswered I see

INVITE - to trunk
then
- TRYING
- RINGING
some of callers CANCEL
OK
ACK

So There is full session too

At CDR I see No Answered session (that is write)
And after that I see empty record file



2014-09-22 18:40 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk:

  THIS IS NOT WHERE YOUR REPLY BELONGS 

 On Monday 22 Sep 2014, Yuriy Gorlichenko wrote:
  2014-09-22 12:12 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk:
   On Monday 22 Sep 2014, Yuriy Gorlichenko wrote:
Hello I have an issue wit MixMonitor. I need to record only answered
calls,
so I set b option for this but calls still recording even call no
answered My asterisk version 12.5.1, at my other servers with older
versions of asterisk (11.8 for example) MixMonitor works fine.
  
   What technology are you using for your outgoing calls?  SIP trunk, IAX
   trunk,
   ISDN, mobile or analogue phone lines?
 
  SIP trunks

 Well, SIP certainly allows for full supervisory information  (analogue
 doesn't, and all calls are deemed answered if the exchange line was
 available).  What have you got in your sip.conf ?  And what does your SIP
 trunk provider have to say on the matter?  (It wouldn't be totally unknown
 for
 a dodgy telco to provide not-entirely-truthful supe.)

 --
 AJS

 Note:  Originating address only accepts e-mail from list!  If replying off-
 list, change address to asterisk1list at earthshod dot co dot uk .

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[asterisk-users] MixMonitor with b option recording all calls

[Yuriy Gorlichenko]

Hello I have an issue wit MixMonitor. I need to record only answered calls,
so I set b option for this but calls still recording even call no
answered My asterisk version 12.5.1, at my other servers with older
versions of asterisk (11.8 for example) MixMonitor works fine.
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[asterisk-users] stasis_app_exec: Stasis app 'MyhApp' not registered

[Yuriy Gorlichenko]

Hello. I tryto use Statis at my dialplan to run my app (a)

When Statis is running from making call ( I dial from softphone some exten
and run dialplan  context  where call Statis(MyApp)) Asterisk responsed:

ERROR[61517][C-0019]: res_stasis.c:852 stasis_app_exec: Stasis app
'MyApp' not registered

How I must Register MyApp
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