Qnewbie

You are right. It looks like the registration got time-out error.

I am not sure what's up...

bubblestar

For registration with SIP providers far away from your Asterisk Server or the lagged respond time.  You might try to give a longer registration time-out allowance.

Admin==>Sip Settings==>TOS under SIP Configuration

Adjust Default Incoming/Outgoing Registration Time as needed.

Qnewbie

It DOES require File Editor to change:
users.conf里的ET263，把registersip=yes改为registersip=no
Now it does not pop-up time-out warning but unregistered

yiucsw

finally got it:
add user.conf registersip=yes改为registersip=no

in Configure Trunk Voip add 10002

in Admin -> SIP -> Misc -> register field add
registersip=yes改为registersip=no
588060123:XXXXX@sip.etelephone.cn:10002/588060123

finally need reboot. for change to user.conf
then the status show registered.

ckleea

一般唔需要改registersip=yes 設定

ckleea

回復 277# Qnewbie


This might be helpful to you.

http://www.switchfin.org/index.p ... &Itemid=54#p720


Some general rules:
1. change the GUI port to some other ports if you want to administer via internet. However, your router should support port redirection
2. strong user name and password for GUI, I believe you know you can take off admin as user name for the GUI
3. for user extension, only you can make is a strong password.
4. if you are ok, change the default incoming port of SIP to some other port number

For iptables/router firewall, you can add a few more
such as we are dropping the connection for a range of IP as we discuss in other threads
add rule to allow only trusted connection.

For iptables  in IP01, I think you may need to back up the rules so that it can review and reload upon rebooting. However, I believe rebooting should be a minimal now when you are at switchfin firmware. My previous experience is more than 3X days.

Qnewbie

回復 291# ckleea


Thanks CK C-hing.

A note for C-hing's method:

1. Normally http.conf
default is bindport=8888
However, we can access it thru port 80.
2. Change the default admin to your "favoritename" in manager.conf, just find&replace [admin] to [favoritename].
3. It is common sense.
4. I do not change 5060&10000-20000. Should be able to change it in sip.conf & rtp.conf.

Iptables in switchfin seems to be a cut version of iptables(or the kernal?). Some commands, like block ip range, is unavailable

ckleea

回復 292# Qnewbie


    The new switchfin firmware 432 or above involved the use of reverse proxy for http server. Hence the http port in http.conf is 8888. The developer said it improves the speed and allows upload of the firmware.

I agree with you that the iptables in switchfin is a trim down version.

bubblestar

In Qnewbie's case, if we apply it in Hong Kong using PCCW, the blocking of port 80 by ISP will be quite useful as we are not allowed to access http thru port 80 here.
We can, then use our desire port as we like.

ckleea

To my understanding, you cannot change the internal GUi port for switchfin firmware after 432. All you can are to port forward a different port at router to port 80 of ip01.

Qnewbie

Tested with siptosis, it is annoying with

[Jan 19 17:17:37] NOTICE[375]: chan_sip.c:8073 sip_reg_timeout:    -- Registration for 'skypetester@dynamic' timed out, trying again (Attempt #28)
[Jan 19 17:17:37] WARNING[375]: chan_sip.c:3089 create_addr: No such host: dynamic
[Jan 19 17:17:37] WARNING[375]: chan_sip.c:8156 transmit_register: Probably a DNS error for registration to skypetester@dynamic, trying REGISTER again (after 20 seconds)

The trunk setting is following:
Type: SIP
Provider name: siptosis
Hostname: dynamic
Username&Authname&Fromuser: skypetester
Fromdomain:
Password: XXXXXXX

Any hints?

yiucsw

Now setting up the SPA3102. Previously I setup the SPA3102 PSTN Line as an extension on the IP01 and eone would call the extension and then get a dial tone to dial out.
According to http://forum.voxilla.com/cisco-l ... asterisk-28807.html
A custom trunk with custom dial string: SIP/$OUTNUM$@300 can work with http authentication. is it possible?
want to kave a better way to handle the SPA3102.

ckleea

回復 296# Qnewbie

How do you set up your siptosis? Windows? Linux?

My setup is like this in sip.conf. No need to use gui to set up

[stsTrunk_01]
username = stsTrunk_01
type = friend
secret = xxxxxxxxxxxxxxxxx
host = 192.168.xxx.xxx
nat = no
dtmfmode = auto
canreinvite = no
port = 5072
qualify = yes
defaultip = 192.168.xxx.xxx
incominglimit = 1
outgoinglimit = 1
call-limit = 1
busylevel =

The above is generated from their script during setup. I have not changed this for I think 2 years. I use a Centos 5.5 server to host the asterisk and skype siptosis application.
This is why stsTrunkconfig charges more.

Qnewbie

回復 298# ckleea


The setup on the Skype side remains the same as it has been set up and run smoothly with asterisk 1.6.2 on ATOM PC.

Now I want to shift to IP01, but I encounter this kind of problem(ET263 is almost the same but I do not use et263 anymore for top up problem). It seems the asterisk has been trimmed in IP01 also(have to say, the asterisk 1.2 to my old TS1-101 did NOT have such problem for SIP registrations).

ckleea

回復 299# Qnewbie


   It is not unexpected for the embedded IPPBX to trim down the functionality. You may try the original method as described to see if it works.