Re: [asterisk-users] Asterisk proxying a REFER

2015-05-04 Thread Luca Pradovera

--
Luca Pradovera
luca.pradov...@gmail.com


Hello,
sorry, I managed to lose the reply amidst the traffic.

What we have here is our application server APP with leg A in AsyncAGI in an 
Adhearsion application, which after some magic dials leg B on the office PBX 
through a configured peer.

Leg B then decides that user C knows more about the subject, and initiates a 
blind transfer to C’s phone using the desk phone that sends a REFER request.
When leg B is hung up as part of the REFER going through, Adhearsion receives a 
Hangup and terminates the call.

There is not much else going on there, our original idea was to put a B2BUA on 
the APP server and to have that “swallow” refers so Adhearsion and the APP 
Asterisk never see it.

Thanks!

Luca


 On Apr 28, 2015, at 19:00, asterisk-users-requ...@lists.digium.com wrote:
 
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 Today's Topics:
 
   1. Re: Asterisk proxying a REFER (Matthew Jordan)
   2. hi list need your help (? ??)
   3. Re: adding area code (Motty Cruz)
 
 
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 Message: 1
 Date: Tue, 28 Apr 2015 07:27:29 -0500
 From: Matthew Jordan mjor...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Asterisk proxying a REFER
 Message-ID:
   CAN2PU+6UYLYFDXnt-XZztBz++8gGmKfdYwHRr84F93OzosV=w...@mail.gmail.com
 Content-Type: text/plain; charset=UTF-8
 
 On Mon, Apr 27, 2015 at 10:36 AM, Luca Pradovera
 luca.pradov...@gmail.com wrote:
 Hello,
 we are using Asterisk with Adhearsion as our application server, with
 another Asterisk box acting as the office PBX, where all office phones are
 registered.
 
 A REFER to transfer calls within the office results in the Adhearsion
 application call being dropped, because the leg between the PBX and the app
 server is terminated by the PBX following the REFER.
 Is there a way to configure Asterisk 11 to proxy a refer across a bridge
 instead of following it, so the application server can follow it instead?
 
 
 Hey Luca -
 
 Unfortunately, there is not a simple or easy configuration setting
 that tells Asterisk to proxy the REFER request through. Generally,
 Asterisk doesn't like proxying anything.
 
 There may still be another way to handle this issue, depending on the
 setup. Can you provide a bit more information about the channels on
 the PBX/Adhearsion server, who sends the REFER request, and what
 happens explicitly in the scenario?
 
 Matt
 
 --
 Matthew Jordan
 Digium, Inc. | Director of Technology
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org
 
 
 
 --
 
 Message: 2
 Date: Tue, 28 Apr 2015 17:19:46 +0300
 From: ? ?? satski...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
 Subject: [asterisk-users] hi list need your help
 Message-ID:
   CAFgS45v=t-qkfktypjhj5yijwoh+d5pqy2jxf3w8yn9ir+b...@mail.gmail.com
 Content-Type: text/plain; charset=utf-8
 
 facing problem with  originating  webrtc calls
 
 
 1-when iam  doing call from webrtc iget ice working
 --- SIP read from WS:91.196.158.205:1466 ---
 INVITE sip:0669197533@77.91.132.9 SIP/2.0
 Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315
 Max-Forwards: 69
 To: sip:0669197533@77.91.132.9
 From: Anton sip:1065@77.91.132.9;tag=5i21qaop43
 Call-ID: ocq4hu8eol3kijsgvt6b
 CSeq: 1465 INVITE
 Authorization: Digest algorithm=MD5, username=1065, realm=77.91.132.9,
 nonce=5152b137, uri=sip:0669197533@77.91.132.9,
 response=446883f3c97a49ea7a9a554a1ba31b6a
 X-Can-Renegotiate: true
 Contact: sip:0momhddj@7cvtd9ihs2e8.invalid;transport=ws;ob
 Content-Type: application/sdp
 Session-Expires: 90
 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
 Supported: timer,ice,outbound
 User-Agent: JsSIP 0.6.26
 Content-Length: 2554
 
 v=0
 o=- 4785391175048354014 2 IN IP4 127.0.0.1
 s=-
 t=0 0
 a=group:BUNDLE audio video
 a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
 m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
 c=IN IP4 192.168.88.26
 a=rtcp:2313 IN IP4 192.168.88.26
 a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host
 generation 0
 a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host
 generation 0
 a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype
 active generation 0
 

Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

2015-05-04 Thread Andrew Martin


- Original Message -
 From: Administrator TOOTAI ad...@tootai.net
 To: asterisk-users@lists.digium.com
 Sent: Friday, May 1, 2015 6:42:38 AM
 Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
 
 Le 01/05/2015 00:05, Andrew Martin a écrit :
  - Original Message -
  From: Administrator TOOTAI ad...@tootai.net
  To: asterisk-users@lists.digium.com
  Sent: Thursday, April 30, 2015 4:43:33 PM
  Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call
  In
 
  I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and
  internal phones are located on the 10.10.32.0/21 LAN subnet. I have many
  internal SIP phones, which appear to be working correctly. I have a few
  external phones (Yealink SIP-T32G or other Yealink model) on
  192.168.32.0/24 which have an OpenVPN client configured on them that
  connects back to the LAN network through a pfSense gateway with OpenVPN
  configured on it.
 
  I faced problems with pfsense -no VPN involved- and finally installed
  siproxd on it. Also set the firewall mode to conservative.
 
  Daniel,
 
  Thanks for the information. Do you have an example or documentation on the
  siproxd configuration that you used?
 
 No, just follow the basis of the parameters given by the package. If I
 remember, SIP use the proxy siproxd and RTP is direct.
 

Looking into it further, in my case it does not appear to be a NATing issue,
since running OpenVPN from pfSense means there's no NATing occurring between
the clients or between the clients and the asterisk server.

Although I was unable to reproduce the problems, I did notice some packet loss
and jitter in sip show channelstats, here is a sample:
Peer Call ID  Duration Recv: Pack  Lost   ( %) Jitter 
Send: Pack  Lost   ( %) Jitter
192.168.32.26446613544@1  00:03:03 94  004238 (97.83%) 0. 
00  000244 ( 0.00%) 0.
192.168.32.385b2ebdc92fd  00:03:03 59  01 ( 1.67%) 0. 
00  91 ( 0.00%) 0.0028

I was unable to find documentation each of these columns, but the high 
percentage
of loss for received packets for 192.168.32.26 seems suspicious. Do these 
statistics
indicate a problem?

Thanks,

Andrew




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