Need urgent help - SIP Trunk Between freepbx and asterisk

Hi Team,

Before posting the issue i have spent 2 days resolving it with no luck in google.
Need Quick help.

I have Asterisk Box

I have setup freepbx both are in same lan.

Both Peers are communicating.

at asterisk box

kocpbx/kocpbx 172.17.11.105 5060 OK (1 ms)

at Pbx box.

kocpbx/kocpbx 172.17.11.208 N 5060 OK (1 ms)

at Extensions.conf i Have.

exten => incomingrejected, 1, Dial(SIP/kocpbx/1002)

At Free PBX i have extension 1002 registered

but when call transferred i get below ErrorLog

<--- Reliably Transmitting (NAT) to 172.17.11.208:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.17.11.208:5060;branch=z9hG4bK304250f6;received=172.17.11.208;rport=5060 From: "10102" ;tag=as2878ab46 To: ;tag=as0ff8d94d Call-ID: [email protected] CSeq: 102 INVITE Server: FPBX-2.10.0rc1(1.8.11) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:172.17.11.208:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.11.208:5060;branch=z9hG4bK304250f6;rport
Max-Forwards: 70
From: “10102” sip:[email protected];tag=as2878ab46
To: sip:[email protected]:5060;tag=as0ff8d94d
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2
Content-Length: 0

<------------->

<— Reliably Transmitting (NAT) to 172.17.11.208:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.17.11.208:5060;branch=z9hG4bK304250f6;received=172.17.11.208;rport=5060
From: “10102” sip:[email protected];tag=as2878ab46
To: sip:[email protected]:5060;tag=as0ff8d94d
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.10.0rc1(1.8.11)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

issue was resolved .

thanks to video

That video is for Asterisk, not FreePBX so if you edited the files on your FreePBX system you will find that FreePBX will simply erase them when you reload.

You posted error traces but none of your configs.

If you had posted your SIP trunk config on FreePBX and you sip.conf from Asterisk someone probably would have helped.

Tagging this as urgent made sure that most people ignored it. This is an open forum, the way to get preferred service is to use paid support.

I have used sip_custom.conf for freepbx.
I am Just 10 days old to asterisk and freepbx.

Tagging this as urgent made sure that most people ignored it. This is an open forum, the way to get preferred service is to use paid support.
I got the message and this is my first post at asterisk community I have found youtube to be help then Google, sticky youtube post will be very helpful for newbies. I felt great setting Up freepbx which never happened to me in linux and solaris. I am so excited and possibility of freepbx. May be some day i could return the favor to the community by supporting the development for now i am happy the iax and sip trunk is working :)

You should not have had to used sip_custom.conf Why could you not setup in FreePBX GUI?

I am sure most of the missed communication is language barrier, English must not be your first language.

You need to supply as much information as you can in your posts please.

Glad yo are liking the project.

Yes English is not my first language.
For testing i have used sip_custom.conf. later i did SIP trunk in GUI.
For now all configuration is working fine.