Dear All,
Can someone help me to connect my kamilio in order to make a PSTN call by rewriting host tp PSTN gateway? I changed my config in a manner to do that...The line is ringing but as soon as I open the line on other side the call will hangup...Please find the piece of code
# RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rewritehost("XX.XX.XX.XX"); force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; }
Regards
Hello,
On 10/25/10 3:49 PM, michel freiha wrote:
Dear All,
Can someone help me to connect my kamilio in order to make a PSTN call by rewriting host tp PSTN gateway? I changed my config in a manner to do that...The line is ringing but as soon as I open the line on other side the call will hangup...Please find the piece of code
# RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rewritehost("XX.XX.XX.XX"); force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; }
if all and only the calls to gateway go through this route, then it is ok. Normally, also in the default config, this route is acalled also for re-INVITEs where you should not change the host part.
To be able to give some hints why the call is hung up immediately, you have to grap the SIP trace, using ngrep, wireshark or tcpdump. I have seen such cases when the codecs in 200ok were not appropriate for caller, so it sends quickly ack and bye.
Cheers, Daniel