返回列表 發帖
回復 5# 角色


   
In your recommended website, they claimed that there is conflict between Skype and Wine.  If you choose to install the Skype static version, it will also make Wine unavailable.

TOP

I am going to install siptosis in my system.  Is it a good idea to install it with a general user instead of using root? How many instance can I make when running the free version?

Thanks

TOP

回復 33# ckleea


   
Following your advice, I reinstall siptosis using root as user.  After installation, I tested my voice with Skype's built-in echo123 in GUI (gnome).  However, only speakers work, microphone did not echo my voice recordings.  

After rebooting the system, I did not see any Skype icon appear on Desktop right corner. Is it normal?

To verify Skype is running, I use the command 'top' in CLI and confirmed it is working.  As I notice from  http://wiki.centos.org/HowTos/Skype that there's a conflict between Skype and Wine (more precisely, between Skype and pulseaudio-libs.i686, on which Wine depends), would be plessed if you could tell how to resolve this?

For info., I have not created dialplan in Asterisk to test with Skype for the time being.

TOP

Thanks for the great dialplan example.

TOP

All dialplan codes are copied and put in place now.  Will test it tomorrow morning.  With the observation using 'top' in CLI environment, Skype has been running for more than 3 hours.  Hope it can last without problem until I make the test.

I recall that you have a teachique to keep Skype running without being interrupted by the program per se.  Lets discuss later for details.  Thanks again for your great advice and assistance.

TOP

I follow SipToSis's tutorial to install in /etc/rc.d/rc.local like this

su -l root -c "/root/siptosis/stsTrunk_linux boot"

I think our installation path is different but it should make no differences.

TOP

回復 35# ckleea


   
Could ckleea c-hing please share the Skype sip trunk settings in sip.conf for reference as well?  I cannot figure out how to initiate the call without callee's info in the dialplan.  Thanks

TOP

回復 43# ckleea


   
In fact, I did the similar settings in sip.conf this afternoon already.  Thanks.

TOP

Both Skype and SipToSis are running.  However, I still have registration problem between Asterisk and SipToSis.  Typing "sip show peers" in CLI return "unreachable" remarks.  

My sip.conf and siptosis.cfg are as below:

sip.conf in Asterisk
  1. [skypetestuser]
  2. username=skypetestuser  ; use same as in brackets above
  3. type=friend
  4. context=default
  5. secret=siptosisregpassword
  6. host=192.168.888.888
  7. port=5070
  8. nat=yes
  9. dtmfmode=auto
  10. canreinvite=no
  11. insecure = port,invite
  12. qualify=yes
  13. defaultip=192.168.888.888
  14. incominglimit=1
  15. outgoinglimit=1
  16. call-limit=1
  17. busylevel=1
複製代碼
siptosis.cfg in /opt/siptosis
  1. #Sample Asterisk registration example - comment out NO registration info above first and uncomment the following.
  2. host_port=5070
  3. contact_url=sip:skypetestuser@192.168.888.888:5070
  4. from_url="skypetestuser" <sip:skypetestuser@192.168.888.888:5060>
  5. username=skypetestuser
  6. realm=asterisk
  7. passwd=siptosisregpassword
  8. expires=3600
  9. do_register=yes
  10. minregrenewtime=120
  11. regfailretrytime=15
複製代碼

TOP

回復 50# ckleea


    I only have one siptosis.cfg file in the siptosis folder.  There is no stsTrunk_01.cfg as mentioned in the directory.  I think these other cfg files are generated only when Trunk Builder is used.

TOP

回復 52# ckleea


    Send to you via email.  Hope you can see the discrepancies.  Thanks

TOP

Wow! Such a long script.  I must work hard to digest the content.  Thanks for your help in advance.

TOP

本帖最後由 bubblestar 於 2011-10-20 23:56 編輯

SipToSis 是否已完全變成免費版呢??  因為收費版本的Download 不見了(被剷走),如果係,真是好消息。

http://www.mhspot.com/sts/siptosis_download.php

TOP

Thanks for the useful information.

TOP

Have you tried installing the free version of stsTrunkbuilder and compare it with your paid version about their functionality?

TOP

返回列表