Problem with SIP traffic

[hrodriguez]

Hi everyone It's my first post, I readed a lot of this in Mr Google but I haven't been able to resolve my problem so, I decided to explain here with the hope that you may be able to help me.

The problem is that when I call to some number, the receptor doesn't listen anything, but I listen all. I talked with Fortigate support and them redirected me to this KB but when I applied this, I has started to get worse. Now I don't listen anything when I call someone

The actual configuration is:

[ul]

One out rule with this parameters:[ul]

Source Interfaz: Internal

Source Addres: Alcatel PBX

Destination Interface: WAN2

Destination Adress: All

Schedule: Always

Service: SIP

Enable Nat

UTM

Protocol Options: default

Enable VoIP: VoIP Profile[/ul]

One in rule with this parameters:[ul]

Source Interfaz: WAN2

Source Addres: all

Destination Interface: Internal

Destination Adress: Alcatel PBX

Schedule: Always

Service: SIP

Enable Nat

UTM

Protocol Options: default

Enable VoIP: VoIP Profile[/ul]

One in rule with this parameters:[ul]

Source Interfaz: WAN2

Source Addres: all

Destination Interface: Internal

Destination Adress: SIP_Proxy_VIP

Schedule: Always

Service: SIP

Enable Nat

UTM

Protocol Options: default

Enable VoIP: VoIP Profile[/ul][/ul][ul]

 One virtual IP with this parameters:[ul]

Name: SIP_Proxy_VIP

External Interface: WAN2

Static Nat

External IP: 0.0.0.0

Mapped IP Addres: IP Alcatel PBX

Port Forwarding

External Service Port: 5060

Map to Port: 5060[/ul]

And the VoIP profile has this edit "VoIP_Profile"             config sip                 set rtp disable                 set register-rate 100                 set invite-rate 100                 set block-long-lines disable                 set block-unknown disable                 set log-violations enable                 set log-call-summary disable             end[/ul]

I don't know what Is happening, I feel like I was in the middle of a desert.

Thank you!!

[helpdesk8]

How was this problem fixed? Please advise.

[songzechun]

sometimes  , when I see the problem like this, I usually del the ALG that session-helper about sip.

you can try it, forward your good news!

like this:

config system session-helper

and del the one which have sip

[NothingKai]

Dear Everyone, my fortigate 100D same error.

I can open port 5060. I can call, but cannot talk, cannot listen. Please help.

I try use CLI console:

Reopen CLI and enter the following commands – do not enter the text after //:

[ol]

config system session-helper

show    //locate the SIP entry, usually 12, but can vary.

delete 12     //or the number that you identified from the previous command.[/ol]

But no see SIP session. 

[HEMEIDY]

I HAVE SAME ISSUE

[Arasu2018]

we have the same issue with fortinet 100e .Phone line getting disconnects automatically during the call. happening not in frequent randomly happening. we tried disabling SIP-ALG but it introduce other problems like No audio so we revert back

[Zulnorain]

Hi,

Did you get it resolved?. I am having the same problem with one of my clients. If you resolved it, can you tell me how you did it? 

[JoergChmiel]

Hi all,

I have exactly the same issue and can't find the issue :( 

Tried already everything ... anyone with a fix?

[beltskyy]

the same problem... reading the topics without any solution...

[RachelGomez123]

Connection issues with your IP phone mean you’re unable to connect to the SIP trunk server.

Softphones which connect and run through a PC may experience this from the computer’s firewall blocking the SIP traffic. A SIP phone not connected to a PC may have a connection issue to the server.

You’ll need to check server traffic to see what items aren’t connecting to the server. To do this, you’ll need a network analyzer. Then you’ll be able to find out if it’s a connection issue.

Another server to SIP phone connection problem is an improperly configured phone. In this case, you’ll need your IT or phone specialist to reconfigure the phone. Once it’s reconfigured the phone should connect without a problem.

Regards,

Rachel Gomez