Issue by dialing to my Voip provider

Hi everyone,

I have an PIAF (Freepbx 2.9) installation which is working fine. The trouble is coming when I am trying to make an outbound call to my VOIP provider. The call is generated from XLITE, 3CX and QUTECOM softphones:

  1. Sometimes (50% of outbound calls) the softphone does not ringing but the cell phone yes. In this case, the audio doesn´t work. In the log I have the next:

    Retransmitting #1 (NAT) to VOIPPROVIDER_IP:5060
    .
    .
    .
    WARNING[9799] chan_sip.c: Retransmission timeout reached on transmission [email protected]_IP:5060 for seqno 102 (Non-critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 32001ms with no response

  2. Sometimes (another 50% of outbound calls) the softphone is ringing ok and the cell phone rings too. In this case, the audio is very good. In the log I have the next:

    Reliably Transmitting (NAT) to LOCAL_IP_ADDRES_SOFTPHONE:5060

I´ve checked the codec settings and the sip settings. What would you think about this? My nat setting is:

nat=yes
externip=XXX.XXX.XXX.XXX (My Public Static IP)
localnet=192.168.0.0/255.255.255.0 (My Local Net - Softphone used)

PS: I´ve tried with two different providers ad the result is the same.

Thanks a lot for your help.

I think you have some time of translation issue with your firewall. It is either loosing or timeing out.

If it has any type of SIP helper or ALG turn it off.

Hi, again with my problem, I´ve found the next:

When the call is successfully established, the SIP messages is:

102 INVITE -> 100 TRYING -> 183 SESSION PROGRESS, etc.

When the call fails, the SIP messages is:

102 INVITE -> (Retransmitting #1 (NAT) to IP_VOIPPROVIDER:5060) -> 102 INVITE -> (Retransmitting #1 (NAT) to IP_VOIPPROVIDER:5060) -> 102 INVITE -> etc…

Again, the issue is random, sometimes the first sip Trace sometimes the second trace.

Thanks a lot.

Same comment as before, some type of networking issue.

Yes, i am thinking in a modem issue. I going to contact to my ITSP for support. Thanks everyone.