Patrick: Patrick Baker wrote: Assuming the phones register to SER one way would be: if ( (lookup("location") | lookup("aliases") | (src_ip==<pstn gwy ip address>) ) { xlog("L_INFO", "\n[SER]: Call to local proxy user. \n"); if (!t_relay()) { sl_reply_error(); xlog("L_INFO", "\n[SER]: Call to local proxy user failed. \n"); }; break; }; Forward from SER to Asterisk based on either RURI value on an INVITE or on a failure as is the case with a call to voicemail. The first situation would require you to define a dialplan such that you can identify which inbound calls to SER need to be forwarded to Asterisk. This is much like what you've already done in the above code. The second situation can be handled using the failure_route in SER. For example if Asterisk voicemail is handled in failure_route[6], assign users the acl value "asterisk", setup the failure route handling in the start of your code, then define the failure route. if (is_user_in("Request-URI", "asterisk")) { t_on_failure("6"); setflag(6); log(1, "[SER]: Flag for Asterisk redirect successful. \n"); }; failure_route[6] { xlog("L_INFO", "\n[SER]: START FAILURE BLOCK #7 Unavailable Asterisk user: Time: [%Tf] Method: <%rm> From uri <%fu> To < %tu> IP source address <%is> R-uri: <%ru> Contact Header: <%ct> \n\n"); if (t_check_status("486")) { prefix("b"); xlog("L_INFO", "\n[SER]: FAILURE BLOCK #7 Phone is busy: Time: [%Tf] Method: <%rm> From uri <%fu> \n\n"); } else if (t_check_status("480")) { prefix("u"); xlog("L_INFO", "\n[SER]: FAILURE BLOCK #7 Phone is unavailable: Time: [%Tf] Method: <%rm> From uri <%fu> \n\n"); } else { prefix("u"); xlog("L_INFO", "\n[SER]: FAILURE BLOCK #7 Phone is unavailable for unknown reason: Time: [%Tf] Method: <%rm> From uri <%fu> \n\n"); } rewritehostport("<asterisk server hostname>:<port on server where sip is listening>"); append_branch(); t_relay_to_udp("<asterisk server hostname>", "<port on server where sip is listening>"); break; } You notice the prefix("b") and prefix("u") statements in the above code. This is so that SER can prefix the users extension with the 'u' or'b' character that Asterisk uses to designate a busy greeting or unavailable greeting should be played. You can add additional status checks like a 302 for call forwarding etc. In Asterisk you need to setup sip.conf which I'm assuming you've already done. In addition you need to define extension rules. For individual mailboxes I use the following. See where the prefixed u & b are used in the pattern matching? exten => _XXXXX,1,VoiceMail2(${EXTEN}) exten => _uXXXXX,1,VoiceMail2(u${EXTEN:1}) exten => _bXXXXX,1,VoiceMail2(b${EXTEN:1}) The same approach works for IVR. I haven't done the agent stuff so your on your own there. If 7700 is your lead number for the IVR then: exten => 7700,1,Goto(mymainmenu,s,1) exten => #,2,Hangup ; Hang them up. [mymainmenu] exten => s,1,Ringing ; 2 seconds of ringback exten => s,2,Answer .... etc. Billing seems to be unique to each site and their acconting model. I'll leave that one for you :-) -Steve

-----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Patrick Baker Sent: Tuesday, March 08, 2005 6:01 AM To: serusers(a)lists.iptel.org Subject: Fwd: [Serusers] SIP to SIP calls & Asterisk Can anyone give some insight on this? ---------- Forwarded message ---------- From: Patrick Baker &lt;patricksbaker(a)gmail.com&gt; Date: Sat, 5 Mar 2005 19:12:43 -0500 Subject: Re: [Serusers] SIP to SIP calls & Asterisk To: Steve Blair &lt;blairs(a)isc.upenn.edu&gt;du>, serusers(a)lists.iptel.org Thanks for the information, but its not set up for multiple domains its setup for multiple contexts in asterisk. every user is listed in the subscribers database in format companyname.user(a)domain.com domain always stays the same. We would like to have a user dial 1000(a)domain.com and have ser connect to the applicable owner of that extension on the same context as the user initiating the call. ex: call from companya.ciscophone(a)domain.com calls 1000(a)domain.com, then ser would translate 1000(a)domain.com into companya.bugetone(a)domain.com without relaying to asterisk.... also please note that the extensions are dynamic and are loaded from a db, so if we wanted to add an extension it wouldnt be viable to do a ser reload and kick all the users off... I imagine the only way to implement this is to grab the context the user is calling from (ex companya) and grab the extension being called tied to the companya extension and return the sip user name... then translate that into the uri form of sip:companya.bugetone@domain.com ... if that makes sense let me know if this is even possible.. its important the domain will always stay the same. Thanks! Patrick On Sat, 26 Feb 2005 07:34:58 -0500, Steve Blair &lt;blairs(a)isc.upenn.edu&gt; wrote: > > Patrick: > > Patrick Baker wrote: > > >Hello all, > > > >I've ran into a dilemma regarding on the call structure is setup for > >my system right now. As of current everything goes through asterisk > >ie > > > >sip user -> ser -> asterisk -> sip user > > > >what I want to try and accomplish is sip user -> ser -> sip user. > > > >I believe this would remove unnecessary load on asterisk servers and > >just connect the call directly. > > > >I'm having a hard time understanding how I will do this thought. as > >of right now I have a forward statement > > > >if (uri=~"^sip:[0-9]*@.*") { > > forward( 10.0.18.3, 5060 ); > > break; > > }; > > > > > > > Assuming the phones register to SER one way would be: > > if ( (lookup("location") | lookup("aliases") | (src_ip==<pstn gwy > ip address>) ) > { > xlog("L_INFO", "\n[SER]: Call to local proxy user. \n"); > if (!t_relay()) { > sl_reply_error(); > xlog("L_INFO", "\n[SER]: Call to local proxy user failed.

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