How to "send required SIP headers" for sip trunk

Asterisk Asterisk General
#1

Hi
im trying to contact a provider for sip trunk whic is plivo and setup a sip trunk.

i could setup the sip trunk to recive calls from my number to my internal extention but i cant make out calls.

after contatcing the support they told me that my ""pbx shoud be sending the required SIP headers “”

and here is a doc for that :
plivo.com/docs/getting-star … oft-switch

for me i dont underdtand that thing !!

i will post my current config for sip trunk & registrer string :

================
type=peer
host=phone.plivo.com
user=phone
fromuser=xxxx
username=xxx
authname=xxx
secret=xxx
fromdomain=phone.plivo.com
context=from-pstn
insecure=port,invite
disallow=all
allow=ulaw

============
my reaister string is :

xxx:xxx>aA@phone.plivo.com:5060/xxxx

agian i can recive incomming calls but i cant make out calls and what i need is some thing i dont undertand it which is
"" sending the required SIP headers"""

can you help me plz ?

#2

You should ask them “and what are the the required SIP headers?”.

If your provdier has a good support you can work that with them in a live session debugging the sip session.

#3

Your link is incomplete.
However: Either Your provider may assist You or - based on a documentation from Your provider - You may request professional support If You can’t get it running yourself …

#4

so do you think that you cant help me ?

#5

Who should be able to help You without any useful documentation (link incomplete, I told You) ?
Hence Your provider requires special SIP-Headers it’s Non-Standard and therefor the exact knowledge of the required headers is necessary to assist You …

#6

thank you all guys , i will check and msg u back with result

cheers

#7

thanks guys i had figured it out.

i created custom dial plan and put in it as the provider told me and it worked

cheers