How to setup reliance JIO trunk (SIP provider in INDIA)

siptrunk
Tags: #<Tag:0x00007fafbeb9a328>

(Lal) #1

Hi,
Dear Community Team members,
How to configure reliance JIO SIP trunk (provider in india)
They are provide following information;
Signal IP: 10.1.1.5
Media IP: 10.1.1.6
Pilat Number: 9999999944
DID Range: 9999999942-43 and 9999999945-65
Username & password is not required

how to setup a sip trunk with the above information, Please find the connectivity diagram
Please suggest a solution (Is there any help doc?)


(Andreas Keller) #2

Hi Lal

create a pjsip trunk - pjsip settings
Authentication = none
Registration = none
SIP Server = 10.1.1.5
SIP Server Port = 5060

#Advanced
From Domain: = 10.0.0.1

#Codecs
include alaw

Help for dial pattern: https://wiki.freepbx.org/display/PHON/Dial+Patterns

Andi


(Lal) #3

Thank you for your reply,
Few minutes back I had configured chan_SIP trunk.
Now all outgoing calls working perfectly, but all incoming calls are respond “all circuits are busy”
My trunk config as follows;

transport=udp
host=XXXXX
fromdomain=XXXXXXXXXXX
username=XXXXXXXXX
secret=XXXXXXXXX
insecure=very
context=from-internal
type=friend
promiscredir=yes
qualify=yes
canreinvite=yes


(Lal) #4

transport=udp
host=XXXXX
fromdomain=XXXXXXXXXXX
username=XXXXXXXXX
secret=XXXXXXXXX
insecure=very
context=from-internal
type=friend
promiscredir=yes
qualify=yes
canreinvite=yes

These configurations are given in Chan_SIP Trunk --> SIP setting Tab–> Outgoing --> Peer details

Incoming tab currently blank. Is there any config required?


(Lorne Gaetz) #5

Looks like this trunk doesn’t use registration, so you must ensure that inbound calls are being sent to the chan_sip port.


(Andreas Keller) #6

why you include secret? As my understanding is, that there is no authentication required.

context from-internal ?

nerveless, disable this trunk and add a pjsip trunk as I recommended. Give pjsip a try!

Andi


(Lal) #7

That update now only given by My provider. i will try and update you


(Lal) #8

Thanks for your reply,
I had configured PJSIP Trunk. Outgoing calls working fine,
incoming calls are connected without audio.

What should I do for getting audio in incoming calls?


(Andreas Keller) #9

seems to me NAT / routing problems.

Can you send a log of a incoming call?

Can you tell me more about your Chan PJSIP settings? (Settings - Asterisk SIP settings)


(Lal) #10

Yes it maybe a NAT issue,
Please find the attached Chan pjsip setting



(Andreas Keller) #11

have you entered the 10.0.0.0/30 ? (Local Networks)

Why 5065 ?


(Lal) #12

yes entered 10.0.0.0 as local network, its entered by click on detect network settings button.

5060 port is used by client iphones (chansip), that why i set this 5065 port here.
pls give me a suggestion for correcting this.


(Andreas Keller) #13

Need trace of a incoming call.

Console
asterisk - r
core set verbose 9

Make a call


(Lal) #14

Thank you for your valuable suggestions. Both incoming and outgoing calls are working now.
The issue (incoming calls) was due to mistakenly configured wan ip in asterisk NAT settings.