Thanks Bogdan for your prompt reply but seems that don't work as expected: i need to strip leading '0' from called R-URI and To !
Just to help, i try to describe better my context: for any external calls, i use route[pstn]: route[pstn] { # Default outbound carrier $var(carrier) = "pstn"; # Need to route to specific carrier ? if(avp_db_load("$fu","$avp(out_carrier)")) { $var(carrier) = $avp(out_carrier); # Remove leading zero subst_uri('/sip:0(.*)@(.*)/sip:\1@\2/g'); subst('/^To:(.*)sip:0(.*)@(.*)/sip:\1@\2/g'); <---- Seems that don't work !!! } # Need to map outbound caller number ? if(avp_db_load("$fu","$avp(out_number_map)")) { uac_replace_from("$avp(out_number_map)","sip:$avp(out_number_map)@$Ri"); append_hf("P-Asserted-Identity: <sip:$avp(out_number_map)@$Ri>\r\n"); } xlog("L_INFO","$ci - Route via $var(carrier) from $fU to $tU (RURI: $ru)\n"); if(route_to_carrier("$var(carrier)")) { t_on_failure("next_gw"); t_relay(); exit; } } Here are dynamic routing tables: dr gateways +----+-----------+------+-------------+-------+------------+-------+------------+-------+--------+--------------------+ | id | gwid | type | address | strip | pri_prefix | attrs | probe_mode | state | socket | description | +----+-----------+------+-------------+-------+------------+-------+------------+-------+--------+--------------------+ | 2 | mediabox1 | 1 | 172.y.x.x | 0 | NULL | NULL | 2 | 0 | | Mediabox gateway | | 1 | pstn1 | 1 | 172.y.x.z | 0 | NULL | NULL | 2 | 0 | | Patton GW to MD110 | | 5 | toip1 | 1 | 172.w.x.r | 1 | NULL | NULL | 2 | 0 | | Trunk VoIP Fastweb | | 6 | toip2 | 1 | 172.w.x.f | 1 | NULL | NULL | 2 | 0 | | Trunk VoIP Fastweb | +----+-----------+------+-------------+-------+------------+-------+------------+-------+--------+--------------------+ dr groups +----+----------+--------+---------+-------------------+ | id | username | domain | groupid | description | +----+----------+--------+---------+-------------------+ | 1 | .* | .* | 1 | PSTN | | 2 | .* | .* | 2 | Asterisk mediabox | | 5 | .* | .* | 3 | Trunk TOIP | +----+----------+--------+---------+-------------------+ dr carriers +----+-----------+-------------+-------+-------+-------+-------------------------+ | id | carrierid | gwlist | flags | state | attrs | description | +----+-----------+-------------+-------+-------+-------+-------------------------+ | 6 | legacy | pstn1 | 1 | 0 | | Carrier to legacy MD110 | | 2 | mediabox | mediabox1 | 1 | 0 | | Carrier to MEDIA BOX | | 1 | pstn | pstn1 | 1 | 0 | | Carrier to PSTN | | 5 | toip | toip1,toip2 | 1 | 0 | | Carrier to Trunk TOIP | +----+-----------+-------------+-------+-------+-------+-------------------------+ dr rules +--------+---------+--------+---------+----------+---------+-------------+-------+-----------------------+ | ruleid | groupid | prefix | timerec | priority | routeid | gwlist | attrs | description | +--------+---------+--------+---------+----------+---------+-------------+-------+-----------------------+ | 1 | 1 | | | 100 | NULL | pstn1 | NULL | Default route to PSTN | | 2 | 2 | | | 100 | NULL | mediabox1 | NULL | Route to MEDIA BOX | | 6 | 3 | | | 100 | NULL | toip1,toip2 | NULL | VoIP Trunk | When someone call 00xxxxxxxx and need to get out via "toip" carrier, just for example, i need to strip out first 0... Thanks, Michele Il 29/04/2016 15:59, Bogdan-Andrei Iancu ha scritto: > Hi Michele, > > the per-gw ops are done in all the routing scenarios (per prefix, per > carrier, etc). Are you sure your call is routed via that GW ? try to > print in cfg the GW ID to see it the right GW is used. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > On 29.04.2016 12:02, Michele Pinassi wrote: >> Hi all, >> >> on my OpenSIPS 1.11.6 i use dymanic module routing to magare multiple >> routes. I need to strip a number for particular gateways and, following >> manual, i set to '1' the 'strip' field in dr_gateways table. >> >> But, using function "route_to_carrier" to manage carrier routing, i get >> no number strip... >> >> Maybe i'm missing something ? >> >> Thanks, Michele >> > -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - central...@unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users