PIONEERS Sipwise Community Edition

hecatae

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I'm currently tinkering with:

Code:
Linux debian 3.16-0.bpo.3-amd64 #1 SMP Debian 3.16.5-1~bpo70+1 (2014-11-02) x86_64
Welcome to Sipwise NGCP platform version 'mr4.0.1'
    _      _    _           
  __| | ___| |__ (_) __ _ _ __ 
/ _` |/ _ \ '_ \| |/ _` | '_ \
| (_| |  __/ |_) | | (_| | | | |
\__,_|\___|_.__/|_|\__,_|_| |_|
                               
System information as of: Wed Sep  9 15:22:33 EDT 2015
 
System load:    0.88    Memory usage:    93.7%
Usage on /:    14%    Swap usage:    68.8%
Local users:    0

Sadly it is running
root@debian:~# asterisk -r
Asterisk 1.4.24.1-RSP (Community supported branch), Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================

Versión RSP de Asterisk 1.4.24.1-RSP (Community supported branch), mantenida por la comunidad
Lista de correo de la versión RSP: http://groups.google.com/group/asterisk-es-rsp
Wiki de la versión RSP: http://www.asterisk-es-rsp.org
Rep. SVN Asterisk-rsp: http://asterisk-es-rsp.irontec.com/svn/asterisk-es-rsp/branches/
Visor web del repositorio SVN: http://asterisk-es-rsp.irontec.com

=========================================================================
Connected to Asterisk 1.4.24.1-RSP (Community supported branch) currently running on debian (pid = 4308)

Though it is behind Kamailio I believe,

Install information is here: https://www.sipwise.org/doc/mr4.0.1/spce/ar01s04.html

You need Debian Wheezy 64bit and at least 1gb of ram
 

ou812

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Have you got this working as in Trunks & users

gary.
 

wardmundy

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Very impressive docs, but Asterisk 1.4. Seriously???
 

hecatae

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Very impressive docs, but Asterisk 1.4. Seriously???


that was my thoughts when I got into the cli, there is a lot of trust in Kamailio to have asterisk on such an old version
 

chris_c_

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Does that mean Sipwise is having the SRTP audio traffic pass thru itself not thru asterisk, except for routing voicemail in and out of asterisk? If true this would mean the modules that deal with audio, STT , TTS, IVR prompts, dial by name, and MOH would need to plug in to Sipwise not Asterisk. That'd be a pretty radical switchover
 

chris_c_

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chris_c_ yes, pretty radical departure, very fun to work with though
Assume you're running Sipwise Community on a virtual machine, behind a SOHO home/office NAT gateway?
When you bring your smartphone/tablet/laptop to a second NAT location, not the one where the server is running (ie coffee shop, hotel lobby, home/office) and try to connect to your sipwise pbx with your SIP dialer app... ie CSipSimple, SipDroid, Zoiper, X-Lite... etc....
How well does Sipwise handle the double-NAT traversal ?
Is your voip app able to Register on Sipwise, make calls, receive calls, and achieve 2-way audio, no problem?
 

hecatae

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Assume you're running Sipwise Community on a virtual machine, behind a SOHO home/office NAT gateway?

I have it on a vps on a public internet address over at cloudatcost
 

chris_c_

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I have it on a vps on a public internet address over at cloudatcost
OK so probably your phone/tablet/desktop is running a sip phone app from behind a NAT gateway?
No problems with one-way audio, or no audio ?
 

chris_c_

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no issues with audio
Would be great to do a test with Sipwise/Kamailio sip proxy, and see if it traverses double-NAT (NAT on both client sip phone, and on the server), and multi-double-NAT (two or more NAT on either the client or server, and NAT on the other).
These are very common situations.
 

ou812

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hecatae@ did you do anything special to get your sip trunk to register, I tried with a Voip.ms trunk that I normally use with Asterisk but was unable to get Sipwise to register, so I only could do outbound calls, I was able to call from user to user internally.
 

hecatae

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hecatae@ did you do anything special to get your sip trunk to register, I tried with a Voip.ms trunk that I normally use with Asterisk but was unable to get Sipwise to register, so I only could do outbound calls, I was able to call from user to user internally.


ou812 I only use ip authentication on the sip trunks I connect to, how do voip.ms request you connect?
 

ou812

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hecatae I am using user name & password, they may have a option to change to Ip authentication I will have to check.

Thanks,
gary.
 

hecatae

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Ok,

tried installing this on my new c@c storage space, so slow on 2gb ram...

fusionpbx is far faster...

off to play with astpp now
 

chris_c_

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any results hecatae with getting video calls to work through sipwise kamailio, under a variety of public nat type situations ie coffee shops, walking on street with smart phone on 3g 4g LTE, ,etc.
 

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