No 180 Ringing after 100 Trying when devices registered with Public IP or Domain Name in Local or private network

Asterisk Asterisk Integration
1

Hello everyone,

I have integrated SIP server where Flexisip as frontend SIP Proxy server and Asterisk as backend SIP Registrar server.
Calls were working fine with 2 way audio in both local and public network in transport TCP.

But I am facing only two issues:
1.calls between Public network to Public network (i.e) both the device registered with Public IP and connected to public network (for example cellular network) is getting disconnected after 32 seconds when the transport is in UDP(especially in UDP Transport).

2.When the two devices registered with Public IP and connected in the Local network either in TCP or UDP transport and trying to call,
the call is not even getting connected. After 100 Trying , there was no 180 Ringing.
But calls were working when devices are connected to cellular network /public network with same registration configuration.

Kindly help me regarding this or share your thoughts which may helps me to resolve the issue.

Thanks in Advance

2

You’d need to actually show a SIP trace and console output.

3

Public_IP_Local_Network_Asterisk_log.txt (698.2 KB)

logs for issue #2 listed above

4

It appears to be in a loop. You call Flexisip, it sends the call back to Asterisk, Asterisk sends the call back to Flexisip, over and over. I have no experience with Flexisip so don’t know anything about it or why it is doing that.

5

Thanks
32Seconds_drop_UDP_public_network.txt (40.6 KB)

This is the logs for issue 1 listed above

6

Asterisk is sending a 200 OK to Flexisip, and not getting an ACK. Once the timer runs out the call is terminated.

7

Thanks for the information,
But its not happening while using TCP as transport.
Is there any way to handle this condition from asterisk side?

8

No. An ACK is required to the 200 OK, it’s part of the SIP specification.

9

Hello guys, kindly apologise for bothering you. May you help me with configuration Flexisip as front-end part and Asterisk as back-end? They are running on different machines with diffrent addresses. I checked the official docs Push Gateway - XWiki but can not figure out how I should point flexisip to asterisk.
I tried specified [module::Forward] route=sip:10.0.40.44:5060;transport=tcp

I already have a running Flexisip instance working in local and public networks. it works okay. But as I understand for frontend proxy server i dont need working sip configuration so I started with an starter config as separate part and finished for now :slight_smile:

10

Do you run Asterisk and Flexisip on the same machine?

11

yes

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