Help with Voice pulse set up

pdang411

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Sep 13, 2008
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#1
Newbbie need help how to set up sip trunk with voice pulse.What should go into PEER Detail ? What should go into user detail and last user string ?. Called voice pulse and tried to load their module but system state missing php. i'm using version 1.21-4 elastix i got the POT lines to dial out and ring to.
 

lek

Guest
#2
We have this in our sip*.conf

[VP-SIP-NYC]
username=AAAAAAAAAAAA
type=peer
sendrpid=yes
secret=XXXXXXXX
qualify=5000
host=nyc.voicepulse.com
fromuser=AAAAAAAAAAAA
dtmfmode=info
disallow=all
canreinvite=no
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=g726
allow=adpcm
allow=lpc10
 

p2ii

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Sep 10, 2008
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#3
I believe the NYC border controller is going away. this the setup I have now.

Trunk name=VP-SIPJFKA

type=peer
context=from-pstn
username=************
secret=***********
host=jfk-primary.voicepulse.com
qualify=5000
allow=all
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes
 

pdang411

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Sep 13, 2008
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#4
Thank you for the help it solve my problem of getting the trunk register. New situation can make outbound but not in bound. Inbound route is set to ring x200 under ring group 600. I have open port 5060 point to elasix server i have Linksys router.
 

hierzu

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Sep 25, 2008
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#5
Having a similar problem and can't seem to get inbound calls going. I can look at the Asterisk CLI and see the call coming in but SS-NOSERVICE will be played instead of passing to my predetermined extensions. Have you had any luck with resolving your inbound issues? If so any suggestions on what you found would be greatly appreciated.
 

reynolwi

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May 5, 2008
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#6
voicepulse has a module built for freepbx that you install and it configures the system for you. I use voicepulse and i just downloaded the freepbx module and it setup everything for me and havent had any problems.
 

pdang411

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Sep 13, 2008
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#7
Thank you I use the Voice Pulse module for free-pbx and it work. it solved the inbound routing
 

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