Re: [asterisk-users] Asterisk 13 WebRTC Status report

[Johan Wilfer]

Den 2015-09-15 kl. 16:52, skrev asandoval...@gmail.com:

Hello Marek! I’ve been running on an issue with my Asterisk 12
configuration for using WebRTC on a LAN environment for about a month! I
really need some help …

My calls from the browser are done fine. I get ringing, they can be
answered and never drop. The thing is that there is no audio on any
side! But I don’t get any error or warning from JavaScript nor the
Asterisk CLI. I’m using Asterisk 12 + jsSIP.

If you could help me solving this I would be eternally greatful  It’s
for my grade project …
These are my files:
sip.conf: http://pastebin.com/kWwXpi4V
http.conf: http://pastebin.com/ZwJWiiwf
SIP debugging for client REGISTER: http://pastebin.com/GNZETtQb
SIP debugging for extension call (Hello-World recording):
http://pastebin.com/0PxjLwBb

I followed these tutorials. If you have any other useful resource, I’d
be glad if you could share it:
http://stackoverflow.com/questions/26254980/websocket-connection-fails-with-asterisk-11
http://blog.gmc.uy/2014/04/asterisk-12-ubuntu-1204-pjproject-srtp.html

Furthermore, if I want to have a local Asterisk configuration, which
should be the IP address for the http.conf + DTLS certificates?? I tried
with localhost but RTP packets redirect to my eth IP.

Thanks in advance!!


In asterisk you have "rtp set debug on" to see if you get rtp packets.
On your client you can start wireshark and look if RTP packets flow in 
both directions.


If you have RTP traffic, maybe you didn't attach the incoming media to 
an audio/video tag in your html. For example:


html: 
In the event-handler for 'addstream' for the call, you have to attach 
the stream to #remoteView.


/Johan

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[asterisk-users] Agents, Members and queues

[Antonio Mario Molina Saorín]

Hi,

I'd like to use queues in Asterisk and I have a few basic questions 
(I'm a newbie) about Queues:

1.- What are the differences between Agents and Members (if any)? 

2.- I want to implemment a small call center and I think the best way it is by 
using dynamic members (agents¿?). In this case, I don't need declare the 
members in agents.conf, is it? So, I'd simply declare them in sip.conf and I'd 
use "add member" asterisk command. Is it Ok or I'm grong?

3.- I'd like to make an application for that members use call center. I've 
thought use some language (java, python...) and work directly with AMI (using a 
telnet connection) in order to send commands and receive and processing events 
to/from asterisk. Is it a correct way?

Thank you very much in advance.

Mario.

-- 
Antonio Mario Molina Saorín
web: http://antonio-mario.com
twitter: @a_mario_molina

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Re: [asterisk-users] Update peer IP address

[Daniel Heckl]

Sebastian,

If I have understood you correctly, the SIP communication is now via NAT 
instead forwarded ports. For safety, it is much better.

I think it is not because of a UDP timeout, but rather because of a NAT 
timeout. For this is "qualify" exactly the right thing to let the NAT port 
opened. 

Daniel

> Am 14.09.2015 um 21:51 schrieb Marie Fischer :
> 
> 
> On 14.09.2015, at 21:58, Sebastian Kemper  wrote:
> 
>> So I got rid of the firewall rule that opened the RTP ports. And then it
>> dawned on me that I don't even need to open the 5060 port. The REGISTER
>> requests established a UDP connection that the kernel's conntrack module
>> was tracking anyway. The only issue was that the REGISTERs occurred only
>> every 480s and the UDP connections were removed after 180s already.
>> 
>> So at first I raised net.netfilter.nf_conntrack_udp_timeout_stream to
>> 500. That worked. But I didn't really want to raise the default. So
>> instead I added "qualify=yes" to the dtag_inbound peer. Now asterisk is
>> sending an OPTIONS request to Telekom every 120s (I raised the frequency
>> from 60 to 120 by setting "qualifyfreq=120" under [general]), which
>> keeps the connection open.
> 
> As far as I understand, raising the UDP session timeout (or lowering the 
> REGISTER timeout, if possible) is actually the better solution. Most Telcos I 
> know don't answer the OPTIONS request anyway and some might object to the 
> traffic overhead.
> 
> -- 
> marie
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[asterisk-users] Realtime Voicemail MWI

[Nick Olsen]

Greetings All, Regarding this archived post. 
http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html
  
 Did anyone ever find an solution to this? I've got a new box running 
13.3.0 with the exact same issue.
  
 For those that don't read the link.
  
 I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, 
These are loaded into asterisk without the mailbox info. Leading to 
"Received SIP subscribe for peer without mailbox" notices. And non-working 
MWI.
  
 Occasionally, It just works. But only on a peer or two at a time. And 
it'll stop working after a few minutes.
  
 Any ideas? Thanks
  
 Nick Olsen
Network Operations  (855) FLSPEED  x106


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