Re: [asterisk-users] Voicemail asking for login

[Israel Gottlieb]

 Does he have the same voicemail context?From: p...@fiberphone.co.nzSent: April 18, 2017 9:43 AMTo: asterisk-users@lists.digium.comReply-to: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Voicemail asking for login  Hi D'ArcyOn 18/04/2017, at 5:17 am, D'Arcy Cain  wrote:One user (that we know of so far) has a different experience.  In that case they are asked for a mailbox number first.  I have tried searching for this issue but nothing seems to apply.  Most discussions are about "*97" vs. "*98".  Can anyone suggest another field of enquiry?Try this:	asterisk -r	core set verbose 10	[get user to trigger fault]	[examine console output, and post to list if still unclear]If you don't solve it yourself, then we'll be able to help further once we've seen the output.HTH,Pete-- 
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Re: [asterisk-users] PBX selection

[J Montoya or A J Stiles]

On Monday 17 Apr 2017, Speed Boy wrote:
>  Hi all, I'm new to VoIP, now we have a project that needs a
>  PBX with client APPs.
> In our team we have argument for choosing PBX. By so far, we
>  have following candidates:
> 
> A: Open source
> 
>  1) Asterisk PBX (http://www.asterisk.org) (with longest
>  history that almost every one knows it, now the last version using the
> PJSIP stack)
>  2) FreeSwitch (http://www.freeswitch.org) (A lot people
>  recommended it to us)
> 
> 
> B: Commercial
> 
> 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> acquired by a HongKong company now
> 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> also includes VoIP SDK, WebRTC and offer rebranding app for free.
> 
> My boss prefers the Open Source PBX since they are free,
> but our CTO prefers the commercial editions, according to
> whom the business PBX has better support, and the
> performance is good, and easy to use - considering our team
> all are new to VoIP/PBX.

Proponents of proprietary solutions always like to say "If an Open Source 
solution breaks, who can you call?"  The answer is, "Any sufficiently-competent 
programmer -- it may be broken, but we have all the pieces".  Whereas if you 
spend money on proprietary software and it breaks, then there is only *one* 
place you can call -- and you'd better hope they are interested to fix your 
problem.

On the other hand, if you could get full Source Code and Modification Rights  
(basically, "everything we could do with a GPL program except distribute 
copies"),  a proprietary solution might not be so bad after all.  But since 
the goal of most proprietary software vendors is to extract money from you and 
maintaining you in a state of perpetual helplessness is highly desirable in 
the course of this, do not expect to get such a deal in real life.
 
> We have did some searching of Asterisk, here are my questions:
> 
> 1. Does the last Asterisk using PJSIP stack ?

Yes.

> 2. Does there has the comparison of PJSIP and reSIProcate, sofia(using by
> FreeSwicth) ?

Not sure about this.  We're still using the original chan_sip driver.

> 3. Is it easy to compile and setup Asterisk?

It's about as easy as compiling anything from Source Code.  Harder than LAME 
MP3 encoder, but easier than the Linux kernel.  If you altered `monop` from 
the BSDgames package to make the streets match your local edition of the game, 
you will have no problem whatsoever with building Asterisk.

If you understand the process of what you are doing -- basically, setting up 
an automated process that will examine your server hardware and software 
configuration  (configure),  choosing which parts of Asterisk you want to 
include  (make menuselect),  compiling the selected human-readable Source Code 
into binary code that the computer can understand natively  (make)  and then 
moving the compiled binary code and configuration files from the Source Code 
folder to where the computer is expecting for them to be  (make install)  then 
you should not have too many problems.

It is always preferrable to compile your own Asterisk to fit your hardware and 
include just the bits you want, rather than rely on anyone else's pre-compiled 
package.

> 4. Which Asterisk version is recommended?

The latest one.

> And does Asterisk support Windows
> ?

You can certainly use Windows softphones to talk to Asterisk, but Asterisk 
itself requires a non-toy underlying operating system.  Ubuntu and CentOS are 
the best-supported Linux distributions.  Asterisk has also been seen working, 
to greater or lesser extents, on Solaris and the BSDs.  But Linux was the 
original development environment  (although one of the two original projects 
that ended up merging and becoming Asterisk, many years ago, was originally 
developed on FreeBSD),  and is what most Asterisk telephonistas know.

Any hardware which is capable of running Windows can, of course, run Linux; 
and usually better.

-- 
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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] Can't compile Asterisk on Ubuntu 16

[Tech Support]

All;

I am trying to build and install certified Asterisk 13.13 cert3 on a
Ubuntu 16.04.2 LTS host without much success. I am getting the following
errors when I try to compile.

 

   [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o

res_pjsip/config_transport.c: In function 'transport_apply':

res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.options[0].level = pj_SOL_TCP();

  ^

res_pjsip/config_transport.c:573:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.options[0].optname = pj_TCP_NODELAY();

  ^

res_pjsip/config_transport.c:574:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.options[0].optval = &option;

  ^

res_pjsip/config_transport.c:575:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.options[0].optlen = sizeof(option);

  ^

res_pjsip/config_transport.c:576:6: error: 'pjsip_tcp_transport_cfg {aka
struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'

   cfg.sockopt_params.cnt = 1;

  ^

/usr/src/asterisk-certified-13.13-cert3/Makefile.rules:149: recipe for
target 'res_pjsip/config_transport.o' failed

make[1]: *** [res_pjsip/config_transport.o] Error 1

Makefile:402: recipe for target 'res' failed

make: *** [res] Error 2

 

Has anyone seen this error before? Any insight at all would be greatly
appreciated.

Thanks;

John V.

 

Tech Support

Tech Support

VoIP Business Solutions

240-215-3479 x325

  supp...@voipbusiness.us

 

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Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

[Jonathan H]

Feel free to take a look at
https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-14-on-Ubuntu.md

Ignore the bit about Voipfone and just skip to the "Install Asterisk" bit.

I've used this same script with Asterisk 12,13 and 14 on Ubuntu 15,16
and 17 so this should work!

Let me know how you get on.

On 18 April 2017 at 13:41, Tech Support  wrote:
> All;
>
> I am trying to build and install certified Asterisk 13.13 cert3 on a
> Ubuntu 16.04.2 LTS host without much success. I am getting the following
> errors when I try to compile.
>
>
>
>[CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o
>
> res_pjsip/config_transport.c: In function 'transport_apply':
>
> res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg {aka
> struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].level = pj_SOL_TCP();
>
>   ^
>
> res_pjsip/config_transport.c:573:6: error: 'pjsip_tcp_transport_cfg {aka
> struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optname = pj_TCP_NODELAY();
>
>   ^
>
> res_pjsip/config_transport.c:574:6: error: 'pjsip_tcp_transport_cfg {aka
> struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optval = &option;
>
>   ^
>
> res_pjsip/config_transport.c:575:6: error: 'pjsip_tcp_transport_cfg {aka
> struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.options[0].optlen = sizeof(option);
>
>   ^
>
> res_pjsip/config_transport.c:576:6: error: 'pjsip_tcp_transport_cfg {aka
> struct pjsip_tcp_transport_cfg}' has no member named 'sockopt_params'
>
>cfg.sockopt_params.cnt = 1;
>
>   ^
>
> /usr/src/asterisk-certified-13.13-cert3/Makefile.rules:149: recipe for
> target 'res_pjsip/config_transport.o' failed
>
> make[1]: *** [res_pjsip/config_transport.o] Error 1
>
> Makefile:402: recipe for target 'res' failed
>
> make: *** [res] Error 2
>
>
>
> Has anyone seen this error before? Any insight at all would be greatly
> appreciated.
>
> Thanks;
>
> John V.
>
>
>
> Tech Support
>
> Tech Support
>
> VoIP Business Solutions
>
> 240-215-3479 x325
>
> supp...@voipbusiness.us
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] PBX selection

[Jonathan H]

On 18 April 2017 at 09:40, J Montoya or A J Stiles
 wrote:
>
> It is always preferrable to compile your own Asterisk to fit your hardware and
> include just the bits you want, rather than rely on anyone else's pre-compiled
> package.

Feel free to take a look at
https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-14-on-Ubuntu.md

Ignore the bit about Voipfone and just skip to the "Install Asterisk" bit.

I've used this same script with Asterisk 12,13 and 14 on Ubuntu 15,16
and 17 so this should work!

Let me know how you get on. And if anyone spots anything wrong on
there, let me know!

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Re: [asterisk-users] PBX selection

[Tzafrir Cohen]

On Mon, Apr 17, 2017 at 10:57:27PM +0800, Speed Boy wrote:
>  Hi all, I'm new to VoIP, now we have a project that needs a
>  PBX with client APPs.
> In our team we have argument for choosing PBX. By so far, we
>  have following candidates:
> 
> A: Open source
> 
>  1) Asterisk PBX (http://www.asterisk.org) (with longest
>  history that almost every one knows it, now the last version using the
> PJSIP stack)
>  2) FreeSwitch (http://www.freeswitch.org) (A lot people
>  recommended it to us)
> 
> 
> B: Commercial
> 
> 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> acquired by a HongKong company now
> 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> also includes VoIP SDK, WebRTC and offer rebranding app for free.
> 
> My boss prefers the Open Source PBX since they are free,
> but our CTO prefers the commercial editions, according to
> whom the business PBX has better support, and the
> performance is good, and easy to use - considering our team
> all are new to VoIP/PBX.

I answered elsewhere[1]. I'll just note one important point from my
reply: Asterisk and FreeSwitch are not PBXs. They are telephony servers.
One application you can build using them is a PBX. You can either
program it yourself or use an existing one (e.g.: FreePBX for Asterisk).
It's not clear from your question which of the two you need.

To me personally the real advantage of open source is not the cost. It
is the ability to tweak, and the control you retain. Right now you are
new to VoIP. But that will soon change.

[1] 
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2017-April/019929.html

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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[asterisk-users] Crashes in jitterbuffer with framedata->timer_interval > 1000

[Richard Kenner]

I had three crashes this morning on a divide-by-zero, for example at
abstract_jb.c:1008 in 14.3.0.

Does this ring any bell to anybody?

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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-18 02:42 AM, Pete Mundy wrote:

Try this:

asterisk -r
core set verbose 10
[get user to trigger fault]
[examine console output, and post to list if still unclear]

If you don't solve it yourself, then we'll be able to help further once
we've seen the output.


I can't see much more than at my previous debug level but here it is 
anyway.  Due to line wrapping I added "<<<" to the end of each line in 
case it is not clear where the actual line endings are.


"Alex Chernyshev" <4164251212> going into voice mail for stocktrans2<<<
-- Executing [stocktrans2@LocalSets:19] Set("SIP/alex-0175", 
"_ACCOUNT=stocktrans2") in new stack<<<
-- Executing [stocktrans2@LocalSets:20] 
VoiceMail("SIP/alex-0175", "stocktrans2@VoiceMail,u") in new stack<<<
   > 0x7f7fea5dc000 -- Probation passed - setting RTP source 
address to 72.143.94.110:28503<<<
--  Playing 
'/var/spool/asterisk/voicemail/VoiceMail/stocktrans2/unavail.gsm' 
(language 'en')<<<
   > 0x7f7fea5dc000 -- Probation passed - setting RTP source 
address to 72.143.94.110:28503<<<
[Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4215 __ast_read: DTMF 
begin '*' received on SIP/alex-0175<<<
[Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4219 __ast_read: DTMF 
begin ignored '*' on SIP/alex-0175<<<
[Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4129 __ast_read: DTMF 
end '*' received on SIP/alex-0175, duration 160 ms<<<
[Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4199 __ast_read: DTMF 
end passthrough '*' on SIP/alex-0175<<<
-- Executing [a@LocalSets:1] Verbose("SIP/alex-0175", 
"stocktrans2 entering mailbox") in new stack<<<

stocktrans2 entering mailbox<<<
-- Executing [a@LocalSets:2] Set("SIP/alex-0175", 
"CDR(userfield)=stocktrans2") in new stack<<<
-- Executing [a@LocalSets:3] VoiceMailMain("SIP/alex-0175", 
"stocktrans2@VoiceMail") in new stack<<<

--  Playing 'vm-login.gsm' (language 'en')<<<
[Apr 18 11:45:49] WARNING[-1][C-0004c47b]: app_voicemail.c:10627 
vm_authenticate: Couldn't read username<<<


Everything looks the same as another one that works except for two 
things.  The one that works doesn't have the "Probation passed" lines. I 
am not sure if that is even part of this call.  The other is the line 
with "Playing 'vm-login.gsm'" in it.  at that point the working one has 
this:


--  Playing 'vm-password.gsm' (language 'en')

Not sure if that's useful information since it just describes the 
original issue - that it asks for a login instead of a password.



--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Crashes in jitterbuffer with framedata->timer_interval > 1000

[Kevin Harwell]

On Tue, Apr 18, 2017 at 1:59 PM, Richard Kenner  wrote:

> I had three crashes this morning on a divide-by-zero, for example at
> abstract_jb.c:1008 in 14.3.0.
>
>
This is quite odd. I took a quick look at the code at that line number and
it appears the divider should never be zero, but somehow it got set to or
initialized to that maybe.


> Does this ring any bell to anybody?
>
>
This does not sound familiar. Please create an issue on the Asterisk issue
tracker [1] and attach a backtrace [2]. Debug logs around the time of the
crash may be helpful too if you have those. Which channel type (chan_sip,
local channel, chan_pjsip) is involved, and how you are enabling the jitter
buffer (dialplan function vs configuration) would be good to know as well.

[1] https://issues.asterisk.org
[2] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Thanks!
-- 

Kevin Harwell
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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[asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Ernie Dunbar]

  
  
Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.

First, here's our technical details:

The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT
router. The router has UDP port 1194 forwarded to our server. This
server also runs our office Asterisk PBX, so there isn't any
networking hardware or firewall between the VPN tunnel and the
Asterisk PBX.

The OpenVPN client is an Asus RT-N66U router, which if I'm not
mistaken, runs a somewhat modified version of Tomato. 

I've got the VPN tunnel working well enough. I can do practically
anything from a computer hooked up to the client router as if I were
in the main office where the server is. But any SIP client I use -
whether it's a hardware SIP phone or a soft phone like Zoiper, can
connect to the Asterisk server without issue. Making calls can work,
accepting calls works, but I only get 1 way voice traffic. I can
hear voice data coming in FROM the Asterisk PBX, but I cannot send
any. 

In my experience with SIP, this usually means a firewall is breaking
the connection from the client phone to the Asterisk server. I just
can't for the life of me find what could be wrong. None of the other
traffic is being blocked. The ipfw firewall on the Asterisk PBX is
extremely open (see below). The firewall on the client router is
turned off, and as far as I can tell, most NAT routers don't even
block outbound traffic in the first place.

I can't see how traffic from the TUN interface on the OpenVPN server
even can be blocked going to another IP address on the same box, but
here are the IPFW rules:

root@ldinfo:/etc/asterisk# iptables -L -n
Chain INPUT (policy ACCEPT)
target prot opt source destination
ACCEPT all -- 192.168.0.0/24 192.168.0.3
ACCEPT all -- 192.168.1.0/24 192.168.0.3
ACCEPT all -- 10.8.0.0/24 192.168.0.3
ACCEPT all -- X.X.X.X 192.168.0.3
ACCEPT all -- 192.168.0.3 X.X.X.X
ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194
REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with
icmp-port-unreachable

Chain FORWARD (policy ACCEPT)
target prot opt source destination

Chain OUTPUT (policy ACCEPT)
target prot opt source destination

Chain POSTROUTING (0 references)
target prot opt source destination

192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server
are on.
192.168.1.0/24 is the network that the remote router is on.
10.8.0.0/24 is the network that the TUN device creates.
X.X.X.X is our datacenter.
192.168.0.3 is the IP address of our PBX.

Any assistance would be greatly appreciated.


  

  


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Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Duncan Turnbull]

-- Original Message --
From: "Ernie Dunbar" 
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 


Sent: 19-Apr-17 10:25:59 AM
Subject: [asterisk-users] SIP connections over OpenVPN connection get 
one-way voice.


Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't 
working *quite* as well as we'd hoped.


First, here's our technical details:

The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT 
router. The router has UDP port 1194 forwarded to our server. This 
server also runs our office Asterisk PBX, so there isn't any networking 
hardware or firewall between the VPN tunnel and the Asterisk PBX.



Asterisk maybe replying from the TUN address which may confuse your sip 
client - if you set the TUN address as a proxy that seems to solve it. 
If asterisk is bound to every address then implicitly it shouldn't 
matter where it replies from, but in the openvpn case it seems to reply 
from a different address to the one it was called on and that can 
definitely fool clients. tcpdump on the tunnel can help you see whats 
happening



The OpenVPN client is an Asus RT-N66U router, which if I'm not mistaken, 
runs a somewhat modified version of Tomato.


I've got the VPN tunnel working well enough. I can do practically 
anything from a computer hooked up to the client router as if I were in 
the main office where the server is. But any SIP client I use - whether 
it's a hardware SIP phone or a soft phone like Zoiper, can connect to 
the Asterisk server without issue. Making calls can work, accepting 
calls works, but I only get 1 way voice traffic. I can hear voice data 
coming in FROM the Asterisk PBX, but I cannot send any.


In my experience with SIP, this usually means a firewall is breaking the 
connection from the client phone to the Asterisk server. I just can't 
for the life of me find what could be wrong. None of the other traffic 
is being blocked. The ipfw firewall on the Asterisk PBX is extremely 
open (see below). The firewall on the client router is turned off, and 
as far as I can tell, most NAT routers don't even block outbound traffic 
in the first place.


I can't see how traffic from the TUN interface on the OpenVPN server 
even can be blocked going to another IP address on the same box, but 
here are the IPFW rules:


root@ldinfo:/etc/asterisk# iptables -L -n
Chain INPUT (policy ACCEPT)
target prot opt source destination
ACCEPT all -- 192.168.0.0/24 192.168.0.3
ACCEPT all -- 192.168.1.0/24 192.168.0.3
ACCEPT all -- 10.8.0.0/24 192.168.0.3
ACCEPT all -- X.X.X.X 192.168.0.3
ACCEPT all -- 192.168.0.3 X.X.X.X
ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194
REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with icmp-port-unreachable

Chain FORWARD (policy ACCEPT)
target prot opt source destination

Chain OUTPUT (policy ACCEPT)
target prot opt source destination

Chain POSTROUTING (0 references)
target prot opt source destination

192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server are 
on.

192.168.1.0/24 is the network that the remote router is on.
10.8.0.0/24 is the network that the TUN device creates.
X.X.X.X is our datacenter.
192.168.0.3 is the IP address of our PBX.

Any assistance would be greatly appreciated.

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Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Sebastian Nielsen]

You need to ensure that traffic to the SIP box is sent to the correct IP. Also 
if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT 
and traffic redirection works as is so the Asus router knows it should send the 
traffic through tunnel and not via WAN.
IMPORTANT: Then you must, in the ASUS RT-N66U make a port forward inwards from 
TUN to the phone client.
I would suggest wiresharking on the client side and see which IP Asterisk 
suggest the client should connect back to. This should be the internal IP of 
the asterisk server as seen from the openvpn server's point of view.
Another important thing: The local network in the Openvpns machine locatiin, 
may NOT have same subnet as the network behind the asus.All these must be 
separate, like:server network: 192.168.1.0/24Openvpn tunnel network: 
192.168.2.0/24Asus network: 192.168.3.0/24
Else you get bizarre routing problems when states appear in the state table.
 Originalmeddelande Från: Ernie Dunbar  
Datum: 2017-04-19  00:25  (GMT+01:00) Till: 'Asterisk Users Mailing List - 
Non-Commercial Discussion'  Rubrik: 
[asterisk-users] SIP connections over OpenVPN connection getone-way voice. 

Hi everyone. I'm having some trouble with an OpenVPN tunnel that
isn't working *quite* as well as we'd hoped.



First, here's our technical details:



The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT
router. The router has UDP port 1194 forwarded to our server. This
server also runs our office Asterisk PBX, so there isn't any
networking hardware or firewall between the VPN tunnel and the
Asterisk PBX.



The OpenVPN client is an Asus RT-N66U router, which if I'm not
mistaken, runs a somewhat modified version of Tomato. 



I've got the VPN tunnel working well enough. I can do practically
anything from a computer hooked up to the client router as if I were
in the main office where the server is. But any SIP client I use -
whether it's a hardware SIP phone or a soft phone like Zoiper, can
connect to the Asterisk server without issue. Making calls can work,
accepting calls works, but I only get 1 way voice traffic. I can
hear voice data coming in FROM the Asterisk PBX, but I cannot send
any. 



In my experience with SIP, this usually means a firewall is breaking
the connection from the client phone to the Asterisk server. I just
can't for the life of me find what could be wrong. None of the other
traffic is being blocked. The ipfw firewall on the Asterisk PBX is
extremely open (see below). The firewall on the client router is
turned off, and as far as I can tell, most NAT routers don't even
block outbound traffic in the first place.



I can't see how traffic from the TUN interface on the OpenVPN server
even can be blocked going to another IP address on the same box, but
here are the IPFW rules:



root@ldinfo:/etc/asterisk# iptables -L -n

Chain INPUT (policy ACCEPT)

target prot opt source destination

ACCEPT all -- 192.168.0.0/24 192.168.0.3

ACCEPT all -- 192.168.1.0/24 192.168.0.3

ACCEPT all -- 10.8.0.0/24 192.168.0.3

ACCEPT all -- X.X.X.X 192.168.0.3

ACCEPT all -- 192.168.0.3 X.X.X.X

ACCEPT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:1194

REJECT all -- 112.220.127.26 0.0.0.0/0 reject-with
icmp-port-unreachable



Chain FORWARD (policy ACCEPT)

target prot opt source destination



Chain OUTPUT (policy ACCEPT)

target prot opt source destination



Chain POSTROUTING (0 references)

target prot opt source destination



192.168.0.0/24 is the network the Asterisk PBX and OpenVPN server
are on.

192.168.1.0/24 is the network that the remote router is on.

10.8.0.0/24 is the network that the TUN device creates.

X.X.X.X is our datacenter.

192.168.0.3 is the IP address of our PBX.



Any assistance would be greatly appreciated.




  

  

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Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Ernie Dunbar]

  
  
On 2017-04-18 03:39 PM, Sebastian Nielsen wrote:

  
  You need to ensure that traffic to the SIP box is sent to the
  correct IP. Also if you use split-tunnel (eg: not redirect-gateway
  def1) you must make sure NAT and traffic redirection works as is
  so the Asus router knows it should send the traffic through tunnel
  and not via WAN.

I'm not that well versed in OpenVPN, but it's worth noting that we
have the `push "redirect-gateway def1 bypass-dhcp"` directive set on
the server. I have two independent DHCP servers on either side of
the VPN, so that the clients are getting their IP addresses for
their appropriate networks - 192.168.0.0/24 on the server side, and
192.168.1.0/24 on the client side. 


  IMPORTANT: Then you must, in the ASUS RT-N66U make a port
forward inwards from TUN to the phone client.


I'll give that a shot, but it will have to wait until tomorrow. :)


  
  
  I would suggest wiresharking on the client side and see which
IP Asterisk suggest the client should connect back to. This
should be the internal IP of the asterisk server as seen from
the openvpn server's point of view.
  
  
  Another important thing: The local network in the Openvpns
machine locatiin, may NOT have same subnet as the network behind
the asus.
  All these must be separate, like:
  server network: 192.168.1.0/24
  Openvpn tunnel network: 192.168.2.0/24
  Asus network: 192.168.3.0/24


I'm pretty sure that I've got this subnet separation in place. If I
didn't cover it in my original post, the network looks like this:

Server network: 192.168.0.0/24
OpenVPN network: 10.8.0.0/24
Asus network: 192.168.1.0/24

The Asterisk SIP registration appears to be responding properly to
this - this is what I see when I do a 'sip show peer' for an Aastra
phone that's connecting through the VPN (Asterisk output is
truncated): 

  ToHost   : 
  Addr->IP : 10.8.0.6:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: FrontDesk1
  SIP Options  : (none)
  Codecs   : (ulaw|alaw)
  Codec Order  : (ulaw:20,alaw:20)
  Auto-Framing :  No 
  Status   : Unmonitored
  Useragent    : Aastra 6731i/3.2.2.1136
  Reg. Contact : sip:FrontDesk1@10.8.0.6:5060;transport=udp



  
  
  Else you get bizarre routing problems when states appear in
the state table.
  
  
  
 Originalmeddelande 
Från: Ernie Dunbar  
Datum: 2017-04-19 00:25 (GMT+01:00) 
Till: 'Asterisk Users Mailing List - Non-Commercial
  Discussion'  
Rubrik: [asterisk-users] SIP connections over OpenVPN
  connection get one-way voice. 


  
  Hi everyone. I'm having some trouble with an OpenVPN tunnel that
  isn't working *quite* as well as we'd hoped.
  
  First, here's our technical details:
  
  The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a
  NAT router. The router has UDP port 1194 forwarded to our server.
  This server also runs our office Asterisk PBX, so there isn't any
  networking hardware or firewall between the VPN tunnel and the
  Asterisk PBX.
  
  The OpenVPN client is an Asus RT-N66U router, which if I'm not
  mistaken, runs a somewhat modified version of Tomato. 
  
  I've got the VPN tunnel working well enough. I can do practically
  anything from a computer hooked up to the client router as if I
  were in the main office where the server is. But any SIP client I
  use - whether it's a hardware SIP phone or a soft phone like
  Zoiper, can connect to the Asterisk server without issue. Making
  calls can work, accepting calls works, but I only get 1 way voice
  traffic. I can hear voice data coming in FROM the Asterisk PBX,
  but I cannot send any. 
  
  In my experience with SIP, this usually means a firewall is
  breaking the connection from the client phone to the Asterisk
  server. I just can't for the life of me find what could be wrong.
  None of the other traffic is being blocked. The ipfw firewall on
  the Asterisk PBX is extremely open (see below). The firewall on
  the client router is turned off, and as far as I can tell, most
  NAT routers don't even block outbound traffic in the first place.
  
  I can't see how traffic from the TUN interface on the OpenVPN
  server even can be blocked going to another IP address on the same
  box, but here are the IPFW rules:
  
  root@ldinfo:/etc/asterisk# iptables -L -n
  Chain INPUT (policy ACCEPT)
  target prot opt source destination
  ACCEPT all -- 192.168.0.0/24 192.168.0.3

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Ernie Dunbar]

  
  
On 2017-04-18 03:38 PM, Duncan Turnbull wrote:

  
  
  -- Original Message --
  From: "Ernie Dunbar" 
  To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'" 
  Sent: 19-Apr-17 10:25:59 AM
  Subject: [asterisk-users] SIP connections over OpenVPN
connection get one-way voice.
   
  
Hi everyone. I'm having some trouble with an
  OpenVPN tunnel that isn't working *quite* as well as we'd
  hoped.
  
  First, here's our technical details:
  
  The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind
  a NAT router. The router has UDP port 1194 forwarded to our
  server. This server also runs our office Asterisk PBX, so
  there isn't any networking hardware or firewall between the
  VPN tunnel and the Asterisk PBX.

 
 
Asterisk maybe replying from the TUN address which may
  confuse your sip client - if you set the TUN address as a
  proxy that seems to solve it. If asterisk is bound to every
  address then implicitly it shouldn't matter where it replies
  from, but in the openvpn case it seems to reply from a
  different address to the one it was called on and that can
  definitely fool clients. tcpdump on the tunnel can help you
  see whats happening
 
  


I think I'll need a bit more detail about how to set the TUN address
as a proxy. Is this done on the OpenVPN server, or at the client
end? I'm also going to tell Asterisk to bind to all IPs and then
restart it when there's no calls in progress, perhaps that's all I
need to do?
  


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Re: [asterisk-users] Voicemail asking for login

[Pete Mundy]

> On 19/04/2017, at 7:58 am, D'Arcy Cain  wrote:
> 
> 
> Everything looks the same as another one that works except for two things.  
> The one that works doesn't have the "Probation passed" lines. I am not sure 
> if that is even part of this call.  The other is the line with "Playing 
> 'vm-login.gsm'" in it.  at that point the working one has this:
> 

Presumably also the line containing 'vm_authenticate: Couldn't read username' 
also doesn't appear in the output on a working mailbox either?

I think that's the place to concentrate your efforts.

It shows shortly after the attempt by VoiceMailMain to enter mailbox 
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?

Can you show the equivalent line from a working mailbox (so we can see if it 
also uses the context 'VoiceMail', or maybe something else instead, like 
'default'?).

Pete



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Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Duncan Turnbull]

Sent from my iPhone

> On 19/04/2017, at 11:43 AM, Ernie Dunbar  wrote:
> 
>> On 2017-04-18 03:38 PM, Duncan Turnbull wrote:
>> -- Original Message --
>> From: "Ernie Dunbar" 
>> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
>> 
>> Sent: 19-Apr-17 10:25:59 AM
>> Subject: [asterisk-users] SIP connections over OpenVPN connection get 
>> one-way voice.
>>  
>>> Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't 
>>> working *quite* as well as we'd hoped.
>>> 
>>> First, here's our technical details:
>>> 
>>> The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT router. 
>>> The router has UDP port 1194 forwarded to our   server. This server 
>>> also runs our office Asterisk PBX, so there isn't any networking hardware 
>>> or firewall between the VPN tunnel and the Asterisk PBX.
>>  
>>  
>> Asterisk maybe replying from the TUN address which may confuse your sip 
>> client - if you set the TUN address as a proxy that seems to solve it. If 
>> asterisk is bound to every address then implicitly it shouldn't matter where 
>> it replies from, but in the openvpn case it seems to reply from a different 
>> address to the one it was called on and that can definitely fool clients. 
>> tcpdump on the tunnel can help you see whats happening
>>  
> 
> I think I'll need a bit more detail about how to set the TUN address as a 
> proxy. Is this done on the OpenVPN server, or at the client end? I'm also 
> going to tell Asterisk to bind to all IPs and then restart it when there's no 
> calls in progress, perhaps that's all I need to do?

Set it as a proxy server in your sip phone client, we found using the tun ip on 
the vpn server works, we keep the actual asterisk address as the sip server and 
use the tun ip as the proxy server

Asterisk is probably already bound to all the addresses netstat -nupl should 
show you the addresses it's listening on for udp, if it says 0.0.0.0 it means 
all addresses

sudo tcpdump -i tun0 -s0 -A udp port 5060

Should show you the sip messages going through the tunnel and you can check the 
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Re: [asterisk-users] Voicemail asking for login

[Victor Villarreal]

Hi Darcy,

What Pete think is correct.

Maybe excecuting the following command at Asterisk console, will help you:

asterisk> voicemail show users

And you will get a list of all mailbox configured in your system. Search
for the user with problems.

Finally, in the Asterisk wiki you can find more info:

https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes

Cheers

El 18 abr. 2017 21:18, "Pete Mundy"  escribió:

On 19/04/2017, at 7:58 am, D'Arcy Cain  wrote:



Everything looks the same as another one that works except for two things.
The one that works doesn't have the "Probation passed" lines. I am not sure
if that is even part of this call.  The other is the line with "Playing
'vm-login.gsm'" in it.  at that point the working one has this:




Presumably also the line containing 'vm_authenticate: Couldn't read
username' also doesn't appear in the output on a working mailbox either?

I think that's the place to concentrate your efforts.
It shows shortly after the attempt by VoiceMailMain to enter mailbox
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?

Can you show the equivalent line from a working mailbox (so we can see if
it also uses the context 'VoiceMail', or maybe something else instead, like
'default'?).

Pete


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Re: [asterisk-users] PBX selection

[Alex Epshteyn]

The solution you choose should be based on many factors which should include 
your business requirements, team's experience, your budget, growth expectations 
and more.

You can choose Asterisk or Freeswitch as a platform and start building on that 
- but it is not simple and being new to VoIP you are likely to make mistakes. 
The "do-it-yourself" approach will some money initially, but will be the most 
expensive option long term - as you will be denying the economy of scale. 
Bringing a "smart programmer" won't help much as you will also create a 
"lock-in". In fact, this could be worse than a dependency created when you use 
a commercial or a known open source solution as while you would still be able 
to get help from the community for the "base" part of your pbx, your custom 
part will be much harder to deal with.

Our company started building Asterisk based PBX in 2002 and Multi Tenant PBX in 
2005 - we do this as our core business and are still finding areas for 
improvement :). As your experience with VoIP is minimal I would side with your 
CTO - you should find a solution high enough in the stack to avoid the 
complexity of building it all yourself.

Good luck,

Alex

-- 

Alex Epshteyn
email: a...@thirdlane.com
web: www.thirdlane.com
phone +1 415.261.6601


- Original Message -
> From: "J Montoya or A J Stiles" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Tuesday, April 18, 2017 1:40:47 AM
> Subject: Re: [asterisk-users] PBX selection
> 
> On Monday 17 Apr 2017, Speed Boy wrote:
> >  Hi all, I'm new to VoIP, now we have a project that needs a
> >  PBX with client APPs.
> > In our team we have argument for choosing PBX. By so far, we
> >  have following candidates:
> > 
> > A: Open source
> > 
> >  1) Asterisk PBX (http://www.asterisk.org) (with longest
> >  history that almost every one knows it, now the last version using
> >  the
> > PJSIP stack)
> >  2) FreeSwitch (http://www.freeswitch.org) (A lot people
> >  recommended it to us)
> > 
> > 
> > B: Commercial
> > 
> > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> > acquired by a HongKong company now
> > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> > also includes VoIP SDK, WebRTC and offer rebranding app for free.
> > 
> > My boss prefers the Open Source PBX since they are free,
> > but our CTO prefers the commercial editions, according to
> > whom the business PBX has better support, and the
> > performance is good, and easy to use - considering our team
> > all are new to VoIP/PBX.
> 
> Proponents of proprietary solutions always like to say "If an Open
> Source
> solution breaks, who can you call?"  The answer is, "Any
> sufficiently-competent
> programmer -- it may be broken, but we have all the pieces".  Whereas
> if you
> spend money on proprietary software and it breaks, then there is only
> *one*
> place you can call -- and you'd better hope they are interested to
> fix your
> problem.
> 
> On the other hand, if you could get full Source Code and Modification
> Rights
> (basically, "everything we could do with a GPL program except
> distribute
> copies"),  a proprietary solution might not be so bad after all.  But
> since
> the goal of most proprietary software vendors is to extract money
> from you and
> maintaining you in a state of perpetual helplessness is highly
> desirable in
> the course of this, do not expect to get such a deal in real life.
>  
> > We have did some searching of Asterisk, here are my questions:
> > 
> > 1. Does the last Asterisk using PJSIP stack ?
> 
> Yes.
> 
> > 2. Does there has the comparison of PJSIP and reSIProcate,
> > sofia(using by
> > FreeSwicth) ?
> 
> Not sure about this.  We're still using the original chan_sip driver.
> 
> > 3. Is it easy to compile and setup Asterisk?
> 
> It's about as easy as compiling anything from Source Code.  Harder
> than LAME
> MP3 encoder, but easier than the Linux kernel.  If you altered
> `monop` from
> the BSDgames package to make the streets match your local edition of
> the game,
> you will have no problem whatsoever with building Asterisk.
> 
> If you understand the process of what you are doing -- basically,
> setting up
> an automated process that will examine your server hardware and
> software
> configuration  (configure),  choosing which parts of Asterisk you
> want to
> include  (make menuselect),  compiling the selected human-readable
> Source Code
> into binary code that the computer can understand natively  (make)
>  and then
> moving the compiled binary code and configuration files from the
> Source Code
> folder to where the computer is expecting for them to be  (make
> install)  then
> you should not have too many problems.
> 
> It is always preferrable to compile your own Asterisk to fit your
> hardware and
> include just the bits you want, rather than rely on anyone else's
> pre-compiled
> package.
> 
> > 4. Which Asterisk version is recommended?
> 
> The latest one.

Re: [asterisk-users] PBX selection

[Roamer2998]

Thanks All.

Thanks Alex, we also tested thirdlane PBX, and comparing it with PortSIP
PBX, Vodia PBX, we hope we can make decision next week.

Best regards,

On Wed, Apr 19, 2017 at 10:05 AM, Alex Epshteyn  wrote:

> The solution you choose should be based on many factors which should
> include your business requirements, team's experience, your budget, growth
> expectations and more.
>
> You can choose Asterisk or Freeswitch as a platform and start building on
> that - but it is not simple and being new to VoIP you are likely to make
> mistakes. The "do-it-yourself" approach will some money initially, but will
> be the most expensive option long term - as you will be denying the economy
> of scale. Bringing a "smart programmer" won't help much as you will also
> create a "lock-in". In fact, this could be worse than a dependency created
> when you use a commercial or a known open source solution as while you
> would still be able to get help from the community for the "base" part of
> your pbx, your custom part will be much harder to deal with.
>
> Our company started building Asterisk based PBX in 2002 and Multi Tenant
> PBX in 2005 - we do this as our core business and are still finding areas
> for improvement :). As your experience with VoIP is minimal I would side
> with your CTO - you should find a solution high enough in the stack to
> avoid the complexity of building it all yourself.
>
> Good luck,
>
> Alex
>
> --
>
> Alex Epshteyn
> email: a...@thirdlane.com
> web: www.thirdlane.com
> phone +1 415.261.6601
>
>
> - Original Message -
> > From: "J Montoya or A J Stiles" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> > Sent: Tuesday, April 18, 2017 1:40:47 AM
> > Subject: Re: [asterisk-users] PBX selection
> >
> > On Monday 17 Apr 2017, Speed Boy wrote:
> > >  Hi all, I'm new to VoIP, now we have a project that needs a
> > >  PBX with client APPs.
> > > In our team we have argument for choosing PBX. By so far, we
> > >  have following candidates:
> > >
> > > A: Open source
> > >
> > >  1) Asterisk PBX (http://www.asterisk.org) (with longest
> > >  history that almost every one knows it, now the last version using
> > >  the
> > > PJSIP stack)
> > >  2) FreeSwitch (http://www.freeswitch.org) (A lot people
> > >  recommended it to us)
> > >
> > >
> > > B: Commercial
> > >
> > > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> > > acquired by a HongKong company now
> > > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> > > also includes VoIP SDK, WebRTC and offer rebranding app for free.
> > >
> > > My boss prefers the Open Source PBX since they are free,
> > > but our CTO prefers the commercial editions, according to
> > > whom the business PBX has better support, and the
> > > performance is good, and easy to use - considering our team
> > > all are new to VoIP/PBX.
> >
> > Proponents of proprietary solutions always like to say "If an Open
> > Source
> > solution breaks, who can you call?"  The answer is, "Any
> > sufficiently-competent
> > programmer -- it may be broken, but we have all the pieces".  Whereas
> > if you
> > spend money on proprietary software and it breaks, then there is only
> > *one*
> > place you can call -- and you'd better hope they are interested to
> > fix your
> > problem.
> >
> > On the other hand, if you could get full Source Code and Modification
> > Rights
> > (basically, "everything we could do with a GPL program except
> > distribute
> > copies"),  a proprietary solution might not be so bad after all.  But
> > since
> > the goal of most proprietary software vendors is to extract money
> > from you and
> > maintaining you in a state of perpetual helplessness is highly
> > desirable in
> > the course of this, do not expect to get such a deal in real life.
> >
> > > We have did some searching of Asterisk, here are my questions:
> > >
> > > 1. Does the last Asterisk using PJSIP stack ?
> >
> > Yes.
> >
> > > 2. Does there has the comparison of PJSIP and reSIProcate,
> > > sofia(using by
> > > FreeSwicth) ?
> >
> > Not sure about this.  We're still using the original chan_sip driver.
> >
> > > 3. Is it easy to compile and setup Asterisk?
> >
> > It's about as easy as compiling anything from Source Code.  Harder
> > than LAME
> > MP3 encoder, but easier than the Linux kernel.  If you altered
> > `monop` from
> > the BSDgames package to make the streets match your local edition of
> > the game,
> > you will have no problem whatsoever with building Asterisk.
> >
> > If you understand the process of what you are doing -- basically,
> > setting up
> > an automated process that will examine your server hardware and
> > software
> > configuration  (configure),  choosing which parts of Asterisk you
> > want to
> > include  (make menuselect),  compiling the selected human-readable
> > Source Code
> > into binary code that the computer can understand natively  (make)
> >  and then
>

Re: [asterisk-users] PBX selection

[Telium Technical Support]

Have a look at xCally from Xenialabs too – they are particularly popular with 
call centers (and still asterisk based).

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roamer2998
Sent: Tuesday, April 18, 2017 11:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] PBX selection

 

Thanks All.

 

Thanks Alex, we also tested thirdlane PBX, and comparing it with PortSIP PBX, 
Vodia PBX, we hope we can make decision next week.

 

Best regards,

 

On Wed, Apr 19, 2017 at 10:05 AM, Alex Epshteyn mailto:a...@thirdlane.com> > wrote:

The solution you choose should be based on many factors which should include 
your business requirements, team's experience, your budget, growth expectations 
and more.

You can choose Asterisk or Freeswitch as a platform and start building on that 
- but it is not simple and being new to VoIP you are likely to make mistakes. 
The "do-it-yourself" approach will some money initially, but will be the most 
expensive option long term - as you will be denying the economy of scale. 
Bringing a "smart programmer" won't help much as you will also create a 
"lock-in". In fact, this could be worse than a dependency created when you use 
a commercial or a known open source solution as while you would still be able 
to get help from the community for the "base" part of your pbx, your custom 
part will be much harder to deal with.

Our company started building Asterisk based PBX in 2002 and Multi Tenant PBX in 
2005 - we do this as our core business and are still finding areas for 
improvement :). As your experience with VoIP is minimal I would side with your 
CTO - you should find a solution high enough in the stack to avoid the 
complexity of building it all yourself.

Good luck,

Alex

--

Alex Epshteyn
email: a...@thirdlane.com  
web: www.thirdlane.com  
phone +1 415.261.6601  



- Original Message -
> From: "J Montoya or A J Stiles"   >
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> mailto:asterisk-users@lists.digium.com> >
> Sent: Tuesday, April 18, 2017 1:40:47 AM
> Subject: Re: [asterisk-users] PBX selection
>
> On Monday 17 Apr 2017, Speed Boy wrote:
> >  Hi all, I'm new to VoIP, now we have a project that needs a
> >  PBX with client APPs.
> > In our team we have argument for choosing PBX. By so far, we
> >  have following candidates:
> >
> > A: Open source
> >
> >  1) Asterisk PBX (http://www.asterisk.org) (with longest
> >  history that almost every one knows it, now the last version using
> >  the
> > PJSIP stack)
> >  2) FreeSwitch (http://www.freeswitch.org) (A lot people
> >  recommended it to us)
> >
> >
> > B: Commercial
> >
> > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> > acquired by a HongKong company now
> > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> > also includes VoIP SDK, WebRTC and offer rebranding app for free.
> >
> > My boss prefers the Open Source PBX since they are free,
> > but our CTO prefers the commercial editions, according to
> > whom the business PBX has better support, and the
> > performance is good, and easy to use - considering our team
> > all are new to VoIP/PBX.
>
> Proponents of proprietary solutions always like to say "If an Open
> Source
> solution breaks, who can you call?"  The answer is, "Any
> sufficiently-competent
> programmer -- it may be broken, but we have all the pieces".  Whereas
> if you
> spend money on proprietary software and it breaks, then there is only
> *one*
> place you can call -- and you'd better hope they are interested to
> fix your
> problem.
>
> On the other hand, if you could get full Source Code and Modification
> Rights
> (basically, "everything we could do with a GPL program except
> distribute
> copies"),  a proprietary solution might not be so bad after all.  But
> since
> the goal of most proprietary software vendors is to extract money
> from you and
> maintaining you in a state of perpetual helplessness is highly
> desirable in
> the course of this, do not expect to get such a deal in real life.
>
> > We have did some searching of Asterisk, here are my questions:
> >
> > 1. Does the last Asterisk using PJSIP stack ?
>
> Yes.
>
> > 2. Does there has the comparison of PJSIP and reSIProcate,
> > sofia(using by
> > FreeSwicth) ?
>
> Not sure about this.  We're still using the original chan_sip driver.
>
> > 3. Is it easy to compile and setup Asterisk?
>
> It's about as easy as compiling anything from Source Code.  Harder
> than LAME
> MP3 encoder, but easier than the Linux kernel.  If you altered
> `monop` from
> the BSDgames package to make the streets match your local edition of
> the game,
> you will have no problem whatsoever with building Asterisk.
>
> If you understand the process of what you are doing -- basically,
> setting up
> an

Re: [asterisk-users] PBX selection

[Jai Rangi]

 Well said Alex

On Tue, Apr 18, 2017 at 7:06 PM Alex Epshteyn  wrote:

> The solution you choose should be based on many factors which should
> include your business requirements, team's experience, your budget, growth
> expectations and more.
>
> You can choose Asterisk or Freeswitch as a platform and start building on
> that - but it is not simple and being new to VoIP you are likely to make
> mistakes. The "do-it-yourself" approach will some money initially, but will
> be the most expensive option long term - as you will be denying the economy
> of scale. Bringing a "smart programmer" won't help much as you will also
> create a "lock-in". In fact, this could be worse than a dependency created
> when you use a commercial or a known open source solution as while you
> would still be able to get help from the community for the "base" part of
> your pbx, your custom part will be much harder to deal with.
>
> Our company started building Asterisk based PBX in 2002 and Multi Tenant
> PBX in 2005 - we do this as our core business and are still finding areas
> for improvement :). As your experience with VoIP is minimal I would side
> with your CTO - you should find a solution high enough in the stack to
> avoid the complexity of building it all yourself.
>
> Good luck,
>
> Alex
>
> --
>
> Alex Epshteyn
> email: a...@thirdlane.com
> web: www.thirdlane.com
> phone +1 415.261.6601
>
>
> - Original Message -
> > From: "J Montoya or A J Stiles" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> > Sent: Tuesday, April 18, 2017 1:40:47 AM
> > Subject: Re: [asterisk-users] PBX selection
> >
> > On Monday 17 Apr 2017, Speed Boy wrote:
> > >  Hi all, I'm new to VoIP, now we have a project that needs a
> > >  PBX with client APPs.
> > > In our team we have argument for choosing PBX. By so far, we
> > >  have following candidates:
> > >
> > > A: Open source
> > >
> > >  1) Asterisk PBX (http://www.asterisk.org) (with longest
> > >  history that almost every one knows it, now the last version using
> > >  the
> > > PJSIP stack)
> > >  2) FreeSwitch (http://www.freeswitch.org) (A lot people
> > >  recommended it to us)
> > >
> > >
> > > B: Commercial
> > >
> > > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> > > acquired by a HongKong company now
> > > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> > > also includes VoIP SDK, WebRTC and offer rebranding app for free.
> > >
> > > My boss prefers the Open Source PBX since they are free,
> > > but our CTO prefers the commercial editions, according to
> > > whom the business PBX has better support, and the
> > > performance is good, and easy to use - considering our team
> > > all are new to VoIP/PBX.
> >
> > Proponents of proprietary solutions always like to say "If an Open
> > Source
> > solution breaks, who can you call?"  The answer is, "Any
> > sufficiently-competent
> > programmer -- it may be broken, but we have all the pieces".  Whereas
> > if you
> > spend money on proprietary software and it breaks, then there is only
> > *one*
> > place you can call -- and you'd better hope they are interested to
> > fix your
> > problem.
> >
> > On the other hand, if you could get full Source Code and Modification
> > Rights
> > (basically, "everything we could do with a GPL program except
> > distribute
> > copies"),  a proprietary solution might not be so bad after all.  But
> > since
> > the goal of most proprietary software vendors is to extract money
> > from you and
> > maintaining you in a state of perpetual helplessness is highly
> > desirable in
> > the course of this, do not expect to get such a deal in real life.
> >
> > > We have did some searching of Asterisk, here are my questions:
> > >
> > > 1. Does the last Asterisk using PJSIP stack ?
> >
> > Yes.
> >
> > > 2. Does there has the comparison of PJSIP and reSIProcate,
> > > sofia(using by
> > > FreeSwicth) ?
> >
> > Not sure about this.  We're still using the original chan_sip driver.
> >
> > > 3. Is it easy to compile and setup Asterisk?
> >
> > It's about as easy as compiling anything from Source Code.  Harder
> > than LAME
> > MP3 encoder, but easier than the Linux kernel.  If you altered
> > `monop` from
> > the BSDgames package to make the streets match your local edition of
> > the game,
> > you will have no problem whatsoever with building Asterisk.
> >
> > If you understand the process of what you are doing -- basically,
> > setting up
> > an automated process that will examine your server hardware and
> > software
> > configuration  (configure),  choosing which parts of Asterisk you
> > want to
> > include  (make menuselect),  compiling the selected human-readable
> > Source Code
> > into binary code that the computer can understand natively  (make)
> >  and then
> > moving the compiled binary code and configuration files from the
> > Source Code
> > folder to where the computer is expecting for them to b

Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-18 08:31 PM, Victor Villarreal wrote:

Maybe excecuting the following command at Asterisk console, will help you:

asterisk> voicemail show users

And you will get a list of all mailbox configured in your system. Search
for the user with problems.


VoiceMail  stocktrans2 Angelica Douglas 12

Definitely there.  In fact, I generate all the configs from a database 
with a script so I would be very surprised if one user was different 
from another.


--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-18 08:17 PM, Pete Mundy wrote:

On 19/04/2017, at 7:58 am, D'Arcy Cain mailto:da...@vybenetworks.com>> wrote:


Everything looks the same as another one that works except for two
things.  The one that works doesn't have the "Probation passed" lines.
I am not sure if that is even part of this call.  The other is the
line with "Playing 'vm-login.gsm'" in it.  at that point the working
one has this:



Presumably also the line containing 'vm_authenticate: Couldn't read
username' also doesn't appear in the output on a working mailbox either?


Exactly.  Since it is not all digits it can't be entered.


I think that's the place to concentrate your efforts.

It shows shortly after the attempt by VoiceMailMain to enter mailbox
'stocktrans2' in context 'VoiceMail'. Does this mailbox exist?


Yes.


Can you show the equivalent line from a working mailbox (so we can see
if it also uses the context 'VoiceMail', or maybe something else
instead, like 'default'?).


"" <6477190146> going into voice mail for alex<<<
-- Executing [alex@LocalSets:19] Set("SIP/thinktel-0181", 
"_ACCOUNT=alex") in new stack<<<
-- Executing [alex@LocalSets:20] VoiceMail("SIP/thinktel-0181", 
"alex@VoiceMail,u") in new stack<<<
--  Playing 
'/var/spool/asterisk/voicemail/VoiceMail/alex/unavail.gsm' (language 
'en')<<<
[Apr 18 11:56:47] DTMF[-1][C-0004c485]: channel.c:4215 __ast_read: DTMF 
begin '*' received on SIP/thinktel-0181<<<
[Apr 18 11:56:47] DTMF[-1][C-0004c485]: channel.c:4219 __ast_read: DTMF 
begin ignored '*' on SIP/thinktel-0181<<<
[Apr 18 11:56:48] DTMF[-1][C-0004c485]: channel.c:4129 __ast_read: DTMF 
end '*' received on SIP/thinktel-0181, duration 280 ms<<<
[Apr 18 11:56:48] DTMF[-1][C-0004c485]: channel.c:4199 __ast_read: DTMF 
end passthrough '*' on SIP/thinktel-0181<<<
-- Executing [a@LocalSets:1] Verbose("SIP/thinktel-0181", "alex 
entering mailbox") in new stack<<<

alex entering mailbox<<<
-- Executing [a@LocalSets:2] Set("SIP/thinktel-0181", 
"CDR(userfield)=alex") in new stack<<<
-- Executing [a@LocalSets:3] VoiceMailMain("SIP/thinktel-0181", 
"alex@VoiceMail") in new stack<<<

--  Playing 'vm-password.gsm' (language 'en')<<<
[Apr 18 11:56:53] WARNING[-1][C-0004c485]: app_voicemail.c:10671 
vm_authenticate: Unable to read password<<<


I hung up before entering the password but it does work when the user 
does it.


--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[Pete Mundy]

> On 19/04/2017, at 4:25 pm, D'Arcy Cain  wrote:
> 
>> Does this mailbox exist?
> 
> Yes.

Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail show users' I 
can't see why the vm_authenticate function is failing to read the username :(

If I were any good at coding in C, I'd probably look inside app_voicemail.c 
around line number 10671 and see if I could determine how it reads the username 
and maybe throw some hacky debug output in there to try and determine at which 
point of that process it's failing. But I'm no good at coding in that language, 
so will have to defer to others to help.

Good work on sending through the console clipping and relevant info. Sorry I 
couldn't resolve it for you.

Anyone else got any other ideas?

Pete




smime.p7s
Description: S/MIME cryptographic signature
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