[Asterisk-Users] newbie asterisk SIP config question (using
VoicePulse Connect!)
Henry Junior
henryjunior at gmail.com
Sat May 28 07:41:38 MST 2005
Greetings,
I am new to all this VoIP stuff and have been having a bit of a hard
time getting my soft phone working as a SIP client thru Asterisk. I
apologize to start off with such a simple question and hope it's ok to
post this and see what others have done.
THE GOOD NEWS:
I have successfully setup Asterisk 1.07 on an OSX machine. The build
is running and working successfully. I am able to use a softphone
client only when I use the settings that VoicePulse gave me which
includes their login/pass/etc. When my asterisk server is running I
can call it and do receive confirmation that it's running.
1st OBJECTIVE
The immediate simple goal is to have asterisk dialout through a SIP client.
CODE SAMPLES:
I was given the following SIP.conf client code and am a bit confused
about all the things I need to do in order to get this working. I
assume this goes in my SIP.conf file but I am unclear (a) if/how I
need to register the client (b) what other code might be required to
set this up (c) how to properly configure the settings for my SIP
client now that just about every configuration changes.
[XLITE1] ;www.xten.com
type=friend
username=ChooseAUsername
secret=ChooseAPassword
context=outbound ; match with the outgoing context in extensions.conf
host=dynamic ; This device needs to register
nat=yes ; X-Lite is behind a NAT router
canreinvite=yes ; Typically set to NO if behind NAT
allow=all ; codec choice: GSM consumes far less bandwidth than ulaw
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
WHAT I HAVE DONE SO FAR:
At the moment I am using the iax.conf and extensions.conf file
examples that are suggested for configuring Asterisk with the
voicepulse connect service:
http://voicepulse.custhelp.com/cgi-bin/voicepulse.cfg/php/enduser/popup_adp.php?p_faqid=67&p_created=1097514602
NEXT STEPS UNCLEAR:
Now I am stuck trying to figure out the next step of 'registering' my
device and testing a SIP client. Any tips about precisely what I need
to do would be greatly appreciated. I've tried a variety of things
but so far none have worked. Voicepulse's strengths definitely don't
rest in its customer service dept thanks for letting me post here. I
appreciate any responses.
Regards,
HJ
More information about the asterisk-users
mailing list