[OpenSIPS-Users] Custom E164 Dialplan on CDRTool
Hello Guys, Im trying to change the E164 rule used by cdrtool to charge my calls (is a test enviroment), i create on library/cdr_generic.php the class E164_Brazil Below the code that i insert on cdr_generic class E164_Brazil extends E164 { function E164_Brazil ($intAccessCode='00', $natAccessCode='55[1-9][1-9]',$CountryCode='',$ENUMtldRegexp=([2-9]{7,})) { $this-regexp_international = /^.$intAccessCode.([1-9][0-9]{5,})\$/; $this-regexp_national = /^.$natAccessCode.([2-9][0-9]{6,})\$/; $this-CountryCode = trim($CountryCode); $this-ENUMtldRegexp= trim($ENUMtldRegexp); } } Just to jusitfy the Brazil country code is 55 and our area codes are [1-9][1-9], later we have the number with 8 digits start from 2 I put on my global.inc datasource (opensips section) the value E164_class = E164_Brazil, i reload cdrtool, i restart cdrtool, but all my calls are not billed (all are on-net to cdrtool) Id *Start Time* *Flow* *SIP Caller* *Location* *Sip Proxy* *Apps* *SIP Destination* *Dur* *Price* *KBIn* *KBOut* *Status* *Codecs* 1Nhttp://ser1.ultra.net.br/CDRTool/callsearch.phtml?cdr_source=opensips_radiuscdr_table=radacct201108order_by=RadAcctIdorder_type=DESCbegin_datetime=1314659280end_datetime=1314744900maxrowsperpage=15action=search# 2011-08-30 09:57:51 on-net MY_USER@MY_SERVER audio 55113544@MY_SERVER 00:05 37.89 92.97 Ok (200) G711u If i try to call with internation prefix (00) i get my call billed correctly Have sombody an idea about where is my mistake ? Thanks for all ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Automatic acceptance of calls for some users
Hi all In my system I have about twenty SIP clients all registered to the same OpenSIPS. Only two of these clients have high priority and should be served as fast as possible. The idea was to implement a logic so that for these two clients I have automatic call acceptance. Other calls from other callers have to be handled as usual (the callee accept the call if wanted). Any hints/links about possible implementations? Thank you in advance, Carmelo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] avp set in branch_route
Hello, If an avp is set in a branch_route, does that avp still exist if a failure_route catches a negative reply and sends the call back into request routing for a serial fork? Or, does the avp go away once that branch goes away? - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] avp set in branch_route
avps are per transaction. A branch is just part of a transaction. Until the transaction is terminated the avp stays active. Be aware that a call's avp is write-able in all branches and assigning a value to an existing avp pushes the new value on the top of the avp array. Calls do not go back to request routing after any reply or failure. Assuming you set to trigger all routing block types as in route below, the order is route, branch route, reply route, failure route. route { . do_routing(1); route(my_stuff); t_relay; exit; } on_branch[tb]{ #each branch runs through here (assuming triggered as above.) after the exit is hit. } on_reply[tr]{ #all replies received hit here but not communication timeouts. #avps written here are not saved unless avpops param is set to } on_failure[tf] { #all avps (except from on_reply if appropriate avpops param is not set) and only pseudo vars from the winning branch are available here. #from here you can kick off another serial/parallel fork. #you can call any custom named route block with route(my_stuff). #Be sure that what is in that route block can be used by the # primary route mode (REQUEST_ROUTE, ONREPY_ROUTE, # FAILURE_ROUTE, etc) it is being called from. Example, you # wouldn't call route(my_stuff) (see below) from an ONREPLY_ROUTE # block because it contains t_relay which can not be used in that route # block mode. Calling from the wrong one will cause config check test # to warn of config error. if(use_next_gw()) { route(my_stuff); } } route[my_stuff] { t_on_reply(tr); t_on_branch(tb); t_on_failure(tf); t_relay(); exit; } On Tue, Aug 30, 2011 at 8:54 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hello, If an avp is set in a branch_route, does that avp still exist if a failure_route catches a negative reply and sends the call back into request routing for a serial fork? Or, does the avp go away once that branch goes away? - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- David Singer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Pre append area code to dialed phone number
Hi All I need to pre append an area code to phone numbers as configuring the clients dial plans is driving me mad. How do I update just the phone number in an invite. The match is complicated as each state in Australia has a different set of prefixes which would require the addition of a different area code. Thanks Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users