- # from file /root/.psip/pjsua.cfg
#
# All lines preceded with a "--" below is a valid config file command
#
# Logging options:
#
--log-file /tmp/psip/app-log
#
#
# Network settings:
#
--local-port 5060
# I think ice is important for maintaining routing information for those with dynamic internet ip addresses
# ice seems to cause problems for may people
# --use-ice
#I think this might only work if you have a gizmo account
# --stun-srv=stun01.sipphone.com
#
#
# Media settings:
#
# using default --clock-rate 12000
--quality 6
# using default --ec-tail 200
# using default --ilbc-mode 20
--rtp-port 4000
#
# Some people have experienced problems with the speex codec
# Uncomment the next three lines to fix
# --dis-codec=speex/32000
# --dis-codec=speex/16000
# --dis-codec=speex/8000
# --add-codec pcmu
#
# User agent:
#
# --max-calls 4
--max-calls 2
#
#
# Buddies: You need to establish an account with a sip provider before you can use this
# --add-buddy=sip:yourname@proxy01.sipphone.com
# --add-buddy=sip:yourname@realsip.com
#
# Account 0:
# --id sip:yourname@proxy01.sipphone.com
# --registrar=sip:proxy01.sipphone.com
# --realm *
# --username=yourname
# --password=password
# --reg-timeout=55
#
# Account 0:
--id sip:fredfinster@sip7.vitelity.net
--registrar=sip:sip7.vitelity.net
--realm *
--username=fredfinster
--password=xxxxxx # get the password from your SIP line provider
--reg-timeout=55
#
# Account 1: The "--next-account" line is required for each additional account
# some where in the front or tail of the file you can hand edit add a buddy. or use PSIP button 'add buddy'
--add-buddy sip:86@sip7.vitelity.net
--add-buddy sip:13604031234@sip7.vitelity.net
--add-buddy sip:15038511234@sip7.vitelity.net
--add-buddy sip:15035917890@sip7.vitelity.net
--add-buddy sip:15416725555@sip7.vitelity.net
--add-buddy sip:fredfinster@sip7.vitelity.net
--add-buddy sip:19712390140@sip7.vitelity.net
2nd look about you can start the PSIP application from the command line in an open RSVT terminal. start ---> Utiltity ----> RXVT terminal
pwd /root
cd /usr/local/psip
pwd /usr/local/psip
ls -l pjs*
# execute the PSIP command manual to see what errors show up.
Code: Select all
./pjsua_custom_03-0.9.0
or
# /usr/local/psip/pjsua_custom_03-0.9.0 --config-file /root/.psip/pjsua.cfg
15:22:43.887 os_core_unix.c pjlib 0.9.0-release for POSIX initialized
15:22:43.887 sip_endpoint.c Creating endpoint instance...
15:22:43.888 pjlib select() I/O Queue created (0xb7b6b098)
15:22:43.888 sip_endpoint.c Module "mod-msg-print" registered
15:22:43.888 sip_transport. Transport manager created.
15:22:43.914 pjsua_app.c Too many arguments specified in cmd line/config file
#
# Note, can not have more than 20 (twenty) --add-buddy lines in the pjsua.cfg file.
#
Testing from the command line, will help to see the hidden error messages
that get lost behind the GUI interface.
I did not setup a proxy or use STUN. I can dial out to regular telephones and when I have PSIP up and running people can call me, because I have a $35 account setup
with Vitelity, a SIP to PSTN provider.
http://www.vitelity.net
Give my SIP address a try for testing your SIP phone Kfpusr.
the telephone number will have to be something like sip:16785551212@xxx,sipphone.com Or whatever your SIP provider gives you.
Using the upper left hand call button.
Some packets might never make it, but for many applications such as VOIP, that's okay; you just hear a bit of noise.
Waiting for your forum comments and PSIP test results via VoIP messages