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Extra softwares in Sailfish using CLI, repositories, etc (117)
to SailfishOS by nieldk - 12 hrs, 38 mins ago -
Firefox with Leste (6)
to Maemo 7 / Leste by teroyk - 1 day, 9 hrs ago - more...
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2009-12-18
, 12:05
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Posts: 72 |
Thanked: 21 times |
Joined on Dec 2009
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#2
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here are what I have
first page:
username 1000@mysipserver.com
pass: password
advanced page
username: 1000 (notice NO server applied)
transport: auto
outbound proxy: mysipserver.com
V in discover public IP
0 in loose routing
Keepalive = auto
keepalive period = auto
V in auto detect stun
I had big problems because of the two places it asks for username. Username one is FULL username incl. server - username 2 is only username...
first page:
username 1000@mysipserver.com
pass: password
advanced page
username: 1000 (notice NO server applied)
transport: auto
outbound proxy: mysipserver.com
V in discover public IP
0 in loose routing
Keepalive = auto
keepalive period = auto
V in auto detect stun
I had big problems because of the two places it asks for username. Username one is FULL username incl. server - username 2 is only username...
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2009-12-18
, 12:12
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Posts: 46 |
Thanked: 5 times |
Joined on Sep 2009
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#3
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What are you guys using for SIP clients?
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2009-12-18
, 12:18
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Posts: 18 |
Thanked: 18 times |
Joined on Oct 2009
@ Barcelona, Spain
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#4
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SIP and VPN go a bit difficult together. You will need to set the 'local-ip-address' so that it uses the right source ip address to get outgoing traffic after you are connected.
I use the following script that executes after my openvpn connection goes up which sets the local-ip-address and enables my sip account which connects to the asterisk server:
and this when the connection goes down (disable the sip account):
You can get the account name by doing
In openvpn you can execute these scripts automatically if you set the up/down parameters.
Also see https://bugs.maemo.org/show_bug.cgi?id=1860
I use the following script that executes after my openvpn connection goes up which sets the local-ip-address and enables my sip account which connects to the asterisk server:
Code:
#!/bin/sh run-standalone.sh /usr/bin/mc-tool update sofiasip/sip/_32130 string:local-ip-address=192.168.8.163 run-standalone.sh /usr/bin/mc-tool enable sofiasip/sip/_32130
Code:
#!/bin/sh run-standalone.sh /usr/bin/mc-tool disable sofiasip/sip/_32130
Code:
/home/user # run-standalone.sh mc-tool list ring/tel/ring spirit/skype/mjbroek0 gabble/jabber/mjbroek_40gmail_2ecom0 sofiasip/sip/_32130
Also see https://bugs.maemo.org/show_bug.cgi?id=1860
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2009-12-18
, 13:45
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Posts: 17 |
Thanked: 11 times |
Joined on Dec 2009
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#5
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@kwek, I did some research and I see that "mc-tool" is part of Libmissioncontrol-utils that seem not to be available for N900 at the moment. Is this correct, or am I missing something?
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2009-12-18
, 14:08
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Posts: 18 |
Thanked: 18 times |
Joined on Oct 2009
@ Barcelona, Spain
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#6
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I think its in the default repository even. Did you try to install it?
http://maemo.org/packages/view/libmissioncontrol-utils/
Code:
apt-get install libmissioncontrol-utils
| The Following User Says Thank You to kwek For This Useful Post: | ||
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2009-12-18
, 14:15
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Posts: 1,096 |
Thanked: 760 times |
Joined on Dec 2008
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#7
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Oh man, SIP over VPN connection. that just sounds painful. Good Luck.
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2009-12-18
, 14:16
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Posts: 17 |
Thanked: 11 times |
Joined on Dec 2009
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#8
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my bad 
moving on in my troubleshooting. Kudos to kwek!
moving on in my troubleshooting. Kudos to kwek!
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2009-12-18
, 15:31
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Posts: 739 |
Thanked: 242 times |
Joined on Sep 2007
@ Montreal
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#9
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Has anyone worked out how to improve sip sound for the other side of the call?
At home my ATA (SPA2102) work great all the time but the N900 seems jerky for the other person (I hear them well though).
It's also the same on the local lan with just the echo test!
Anyone had this issue and managed to fix it?
At home my ATA (SPA2102) work great all the time but the N900 seems jerky for the other person (I hear them well though).
It's also the same on the local lan with just the echo test!
Anyone had this issue and managed to fix it?
I've a follwoing problem. In my home network I have test SIP gateway (say 192.168.1.10) that accepts all connections from internal network. I have an account there (say 1000, with 1000 as a password).
I managed to VPN into my home network and can access all its resources while connected over VPN.
However I have a hard time configuring SIP connection on N900. I have tried several combiantions of settings and none of them worked: every time I ended up with "signing in..." window and then "not connected".
Here's what I have tried:
user name: 1000 and 1000@192.168.1.10)
pwd: 1000
transport: auto, tcp
outbound proxy: blank, 192.168.1.10, sip:192.168.1.10
discover public address: on/off
loose routing: on/off
If I vpn using my laptop and configure any software SIP phone applying minimum settings (user 1000@192.168.1.10, pwd 1000, sip server 192.168.1.10, port 5060, transport TCP) it works.
What could I do to troubleshoot this? Are there any ways to check logs of sip module attempting to register itself?
Thanks in advance,
Alex