Hi All,
I hope someone can help me with the following questions I have.
1. I have set up accounting module for record the call, simple outgoing call and incoming call are ok. However, if a sip phone activated the call forward, the call record doesn't show which sip client forward the call. e.g. A call B and B is the sip phone which forward the call to C. In the record, I can only see A call C. How can I see a record that B call C instead of A call C in the record?
2. I have connected to a PSTN gateway, and i have some internal sip users which can make a pstn call, the sip proxy will send their own private number a caller id to the pstn gateway. Since the callid is a invalid number, many Mobile phone networks provider reject the call. I want to remove the from callid when the call being send to the pstn gateway, how can i do this? If someone can send me the code that will be great.
Hi all,
I wish you first a happy new year !! One again, I ask you some help. Thanks a lot for your patience and your answers that really helped me.
So, I want to put a SER in front of one Asterisk for the moment (more after). All Servers have public IP address and so, I don't care about the RTP/SDP messages. All phones are behind NAT somewhere in Switerland :)
I have the configuration below in openser:
modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") #modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 2)
modparam("usrloc", "db_mode", 0) #modparam("rr", "enable_full_lr", 1)
route{
xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n");
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; };
if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; };
# if (search("User-Agent:.*Thomson.*")) { # };
if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { if(method=="OPTIONS") { }; avp_pushto("$ru","$tu"); forward();
}else { fix_nated_contact(); force_rport(); if(method=="REGISTER"){ ds_select_dst("4", "0"); t_relay(); } else { ds_select_dst("0", "4"); forward(); } } }
And in my dispatcher.list I have:
0 sip:212.xxx.xxx.153:5060 4 sip:212.xxx.xxx.153:5060
With this configuration, a snom phone, a X-lite phone and a Cisco phone seams to work .... But with the Thomson ST2030, it doesn't work. In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION message to the FW IP adress and the right port where the thomson is and then, after a while, the FW close the connection. The thomson phone is so "UNREACHABLE" on asterisk status and nobody can call it. I'm not sure that my configuration is ok in ser but I think that is a problem with the Thomson2030. Do you had the same kind of problem or do you understand the problem here ?
Thanks for your help,
Thomas Deillon
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Howard Tang -
Thomas Deillon