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[asterisk-users] Hello again


 
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aaberga at gmail.com
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PostPosted: Fri Sep 30, 2016 6:07 am    Post subject: [asterisk-users] Hello again Reply with quote

Hi,

after a long pause (Asterisk 1.8 times), I have started again playing with VOIP. A lot has changed since last time I did setup an Asterisk system!

So I am asking for some help.

++++++++++++++++

PJSIP seems tougher..

So my problem is that I do have a test system up in the cloud, behind a firewall. I am trying to make the “Hello World!” mandatory call between two iPhones (with the Bria SIP client).

Outcomes are erratic.

================

This is the pjsip.conf file:

——————————————————————

[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.2.12.3/32
local_net=127.0.0.1/32
external_media_address=10.2.12.2
external_signaling_address=10.2.12.2

;===============Messagenet TRUNK

[messagenet_reg]
type=registration
transport=transport-udp-nat
outbound_auth=messagenet_auth
server_uri=sip:xxxxxxxxx@sip.messagenet.it:5061
client_uri=sip:xxxxxxxxx@sip.messagenet.it:5061

[messagenet_auth]
type=auth
auth_type=userpass
password=xxxxxxxx
username=xxxxxxxx

[messagenet_aor]
type=aor
contact=sip:sip.messagenet.it:5061

[messagenet]
type=endpoint
transport=transport-udp-nat
context=messagenet_incoming
disallow=all
allow=ulaw
allow=alaw
outbound_auth=messagenet_auth
aors=messagenet_aor

[messagenet_id]
type=identify
endpoint=messagenet
match=sip.messagenet.it

;===============Extension 2102

[2102]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2102
aors=2102
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no


[auth2102]
type=auth
auth_type=userpass
password=xxxxxxxxxxxxxx
username=2102

[2102]
type=aor
max_contacts=1

;===============Extension 2103

[2103]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2103
aors=2103
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no

[auth2103]
type=auth
auth_type=userpass
password=xxxxxxxxxxxxxx
username=2103

[2103]
type=aor
max_contacts=1

================================

This is a trace of what I do see from the console.

First I let the Bria clients connect. Then I try to call terminal 1 from terminal 2. Most of the times there is no route to the destination, even if it appears as an online AOR (whatever that means!! Ahhh: Good olde times of Peer, Friend, etc… ;-)

A couple of times I got a connection, with the typical one side only audio of NAT traversal problems.

BTW: The iPhones are behind TWO nats (one is given by the broadband router, one by the WiFi router that gives a better WiFi cover for in-house things).

My understanding is that I did something wrong in letting the phones ‘register’ them as present and available to receive calls.

If only I knew what is wrong… I have tried random combinations of rtp_symmetric, force_rport, and friends; nothing final discovered...

====

Thanks in advance for any help,
Aldo


PS: Setting up Zoiper as IAX client works of course like a charm on the VOIP side. The only catch is that Zoiper has less than optimal background support on IOS… And I have no plan to make an IAX client myself!

I want to get my old Asterisk apps back online and the VOIP client part makes no sense to me..


--
_____________________________________________________________________
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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joan.ayma at ackstorm.es
Guest





PostPosted: Fri Sep 30, 2016 7:22 am    Post subject: [asterisk-users] Hello again Reply with quote

Your external addres seems wrong. As you are doing natting, you need to
set to you external (natted behind firewall).


El 30/09/16 a les 13:07, aaberga/gmail ha escrit:
Quote:
Hi,

after a long pause (Asterisk 1.8 times), I have started again playing with VOIP. A lot has changed since last time I did setup an Asterisk system!

So I am asking for some help.

++++++++++++++++

PJSIP seems tougher..

So my problem is that I do have a test system up in the cloud, behind a firewall. I am trying to make the “Hello World!” mandatory call between two iPhones (with the Bria SIP client).

Outcomes are erratic.

================

This is the pjsip.conf file:

——————————————————————

[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.2.12.3/32
local_net=127.0.0.1/32
external_media_address=10.2.12.2
external_signaling_address=10.2.12.2

;===============Messagenet TRUNK

[messagenet_reg]
type=registration
transport=transport-udp-nat
outbound_auth=messagenet_auth
server_uri=sip:xxxxxxxxx@sip.messagenet.it:5061
client_uri=sip:xxxxxxxxx@sip.messagenet.it:5061

[messagenet_auth]
type=auth
auth_type=userpass
password=xxxxxxxx
username=xxxxxxxx

[messagenet_aor]
type=aor
contact=sip:sip.messagenet.it:5061

[messagenet]
type=endpoint
transport=transport-udp-nat
context=messagenet_incoming
disallow=all
allow=ulaw
allow=alaw
outbound_auth=messagenet_auth
aors=messagenet_aor

[messagenet_id]
type=identify
endpoint=messagenet
match=sip.messagenet.it

;===============Extension 2102

[2102]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2102
aors=2102
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no


[auth2102]
type=auth
auth_type=userpass
password=xxxxxxxxxxxxxx
username=2102

[2102]
type=aor
max_contacts=1

;===============Extension 2103

[2103]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2103
aors=2103
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no

[auth2103]
type=auth
auth_type=userpass
password=xxxxxxxxxxxxxx
username=2103

[2103]
type=aor
max_contacts=1

================================

This is a trace of what I do see from the console.

First I let the Bria clients connect. Then I try to call terminal 1 from terminal 2. Most of the times there is no route to the destination, even if it appears as an online AOR (whatever that means!! Ahhh: Good olde times of Peer, Friend, etc… ;-)

A couple of times I got a connection, with the typical one side only audio of NAT traversal problems.

BTW: The iPhones are behind TWO nats (one is given by the broadband router, one by the WiFi router that gives a better WiFi cover for in-house things).

My understanding is that I did something wrong in letting the phones ‘register’ them as present and available to receive calls.

If only I knew what is wrong… I have tried random combinations of rtp_symmetric, force_rport, and friends; nothing final discovered...

====

Thanks in advance for any help,
Aldo


PS: Setting up Zoiper as IAX client works of course like a charm on the VOIP side. The only catch is that Zoiper has less than optimal background support on IOS… And I have no plan to make an IAX client myself!

I want to get my old Asterisk apps back online and the VOIP client part makes no sense to me..



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Departamento de SAT
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--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference

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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asterisk_list at earth...
Guest





PostPosted: Fri Sep 30, 2016 7:49 am    Post subject: [asterisk-users] Hello again Reply with quote

On Friday 30 Sep 2016, aaberga/gmail wrote:
Quote:
Hi,

after a long pause (Asterisk 1.8 times), I have started again playing with
VOIP. A lot has changed since last time I did setup an Asterisk system!

So I am asking for some help.
[stuff deleted]
Quote:
[2102]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2102
aors=2102
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no

You might want to comment out all references to g729 (which needs a special
licence) and just use alaw (the native codec of the PSTN) throughout.

If one of the phones is deciding to use g729 and your Asterisk doesn't have
the relevant licence, then you might well get all manner of strange things
happening.

Even if you have g729 licences, try and get it working with alaw first. The
fewer things there are that could go wrong, the better. It's always best to
get it working with the simplest possible setup first, and only then add
sophistication.

--
AJS

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

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference

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
Back to top
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