search for: t_relay

...ops","avp_url","mysql://root:passwd@192.168.2.75/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops","avp_aliases","inv=i:15") ............... route { ...... if (loose_route()) { t_relay(); exit; }; if(is_method("INVITE") && !has_totag()) { xdbg("user [$ruri] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri","$a...

...if (lookup("location")) { log (1, "******* IP to IP call *************"); if (method == "INVITE"){ setflag (1); t_on_failure("1"); t_relay(); sl_send_reply ("180", "Ringing"); setflag (1); break; } if (!t_relay()) { sl_send_reply("404", "Not Found");...

...? I've follow the SER, Asterisk and Lucent TNT by Michael Shuler (http://www.voip-info.org/wiki-Asterisk+at+large ), but I think I've missed something. The * and ser are using public ip, no nat for them. I've tried different config, with and without rtpproxy, with forward instead of t_relay, but same or more problems. If someone could help me please. Here are my conf files : extensions.conf: ============ [ser] exten => 0870441067,1,Dial(SIP/${EXTEN}@ser,20,t) exten => 0870441067,2,Congestion exten => 0870441067,3,Hangup exten => 0870441111,1,Dial(SIP/${EXTEN}@ser,20,...

...record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method == "REGISTER") record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize(&quo...

...------------------------------------------------------------------------------------------------- if (method == "INVITE") { if (uri =~ "sip:1[0-9]{4}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xx.xx.xx.xx:xxxx"); t_relay(); break; } } ----------------------------------------------------------------------------------------------------------- inside sip.conf i have ----------------------------------------------------------------------------------------------------- register => sipphonenumber:p...

...ig"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; # native SIP destinations...

...tested by me personally. numbers@freenum.org # catch voice:sip ENUM entries, only if PSTN not forced if (enum_query("e164.arpa.", "voice")) { if (is_uri_host_local()) { route(5); break; }; if (!t_relay()) { xlog("L_ERR", "%is [%Tf]: %rm %fu -> %ru [R4]: ENUM destination: relaying failed\n"); sl_reply_error(); break; }; break; }; #start # look up freenum.org ENUM entries i...

...i guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf ------------------------------------------------------------------------------------- port=5061 bindaddr=0.0.0.0 srvlookup=yes [ser] type=peer host=xxx.xxx.xxx.xxx context=ser1 inside extensions.conf -------------...

...ER is running in three IPs -global, private and IPv6 and Asterisk is also running on the same Linux box with different private IP from SER. Here, I have a question regarding relay from SER to Asterisk. Does Asterisk SIP channel should listen to global IP? By that I mean I don't understand how t_relay works. Say, when my party members make connection from different global IPs to my SER on global IP, is that possible to relay that connection to Asterisk running in the same Linux box with private IP and different port number? On Global Ips Linux Box running two servers ___...

...# retrieve voicemail # if (uri=~"^sip:2[0-9]*@magnum.test.net") { log(1, "Retrieving voicemail\n"); # redirect now! rewritehostport("202.125.25.102:5061"); append_branch(); t_relay_to_udp("202.125.25.106","5061"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; timeout occurred ... now to forward to Asterisk...

...of a failure the standby asterisk takes over the IP of the 1st asterisk so the services continues (sync the conf file with rsync for example), however if the phones use a host=dynamic they wont be able to be called until they have reregistered themselves at the backup asterisk. Is there a SER like t_relay kinda thingy to let the backup know the locations of the Sip Phones? Kind regards, E. Versaevel

..., p->fromdomain); /* Send MWI */ After this patch is applied, the MWI NOTIFY messages coming from asterisk will have the URI user@ser.server.tld. This can be then routed with ser to the correct phone with normal SER routing rules. ie. SER does a lookup("location") and then a t_relay(). I don't believe this patch should effect any non-ser controlled sip phones. For me, this method was a lot easier then Method 2 listed above. You can add as may mailbox's as you like into the mailbox= line in the asterisk sip.conf file. One possible problem is if you have a mailbox calle...

...-noanswer-/')){ log(1,"Err in subst_user\n"); } xlog("L_ERR", "Relaying to voicemail No answer\n"); } rewritehostport("FEATURE:5060"); append_branch(); t_relay(); } } >From the gateway's point of view, the invite looks like 1. U GATEWAY:5060 -> PROXY:5060 2. INVITE sip:DSTNUM@PROXY SIP/2.0. 3. Via: SIP/2.0/UDP GATEWAY:5060;branch=z9hG4bK6e117757. 4. From: "SRCNUM" <sip:SRCNUM@GATEWAY>;tag=as7f56ca42. 5. To: <sip:DSTNUM@P...

I encountered this strange problem which is I can call into Asterisk box but I cannot call out. I was successful before using exactly the same euroISDN line but with TE110 and different versions of Asterisk. This time, I am using: . TE410 . Asterisk 1.4.13 . Zaptel 1.4.6 . Libpri 1.4.2 1) I put the following into extensions.conf to get to the outside line exten => 0,1,Dial(Zap/1) 2)

...# timeout occurred ... now to forward to Asterisk's voicemail service if(method == "INVITE" && isflagset(4)) { t_on_failure("1"); }; }; route(1); } # ------------------------------- # Route Processing # ------------------------------- route[1]{ if(!t_relay()){ sl_reply_error(); }; } route[2]{ if(!save("location")){ sl_reply_error(); } } # voicemail activation!! # failure_route[1] { log(1,"Activating voicemail!!\n"); forward(202.122.25.106, 5061); } --------------------------- -------- -------- AST...

Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and

>When I call a SIP user, the phone should ring in more than one >extentions. Also more than one phone should be able to register with >asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for