[OpenSIPS-Users] Callpickup scenario help

HI to all, i need a little help regarding one scenaria (What would be the best way to implement this). I am having opensips which is heandling registration and load balancing. Behind opensips I am having FS servers. What would be the best way to implement group call pickup. I need to know to

Re: [OpenSIPS-Users] memory consumed by t_relay

Hi Bogdan-Andrei, When you get the memory error, could you please do : "opensipsctl fifo get_statistics all" and send me the output (off list) please? When about 10,000 calls, 10kcalls.log . When about 20,000

Re: [OpenSIPS-Users] Registrar not saving received from Path header

Hello Bogdan, Here is the output from the opensips log. I have also attached a snippet from the log file. May 21 23:39:15 OpenSips[14397]: [ID 257313 local1.debug] DBG:registrar:save_aux: xXx - flags param is May 21 23:39:15 OpenSips[14397]: [ID 154992 local1.debug] DBG:registrar:save_au

[OpenSIPS-Users] B2BUA Segfault

Hello, While using the B2BUA module in OpenSIPS 1.9.0, we've encountered a consistent segfault. We are using a refer scenario just like the one in the B2BUA sample docs, and after several REFERs for the same call (to different destinations), OpenSIPS crashes with a segfault. The core file indicate

Re: [OpenSIPS-Users] Do_Routing problem

Hi, I set in dr_rules #lebanon as a carrier route to terminate the rule that match 777961 , Please find below the output for your script loadmodule "drouting.so" modparam("drouting", "db_url", "virtual://set1") # CUSTOMIZE ME modparam("drouting", "use_domain", 1) modparam("drout

Re: [OpenSIPS-Users] Slight problem routing 100s and 183s

Bogdan I am so sorry!!! 192.168.2.11 is actually a UAC polycom phone. The only asterisk box that is being used in the scenario right now is 192.168.2.10, as seen in the traces. Please forgive me! :) N. On 5/21/13, Bogdan-Andrei Iancu wrote: > Hello Nick, > > To be honest, I'm a bit confused - lo

Re: [OpenSIPS-Users] Slight problem routing 100s and 183s

Hello Nick, To be honest, I'm a bit confused - looking at the trace, I see the INVITE comes from .11 (an aterisks), goes to .5 (opensipsIn) and then to .10 (another asterisk)This does not match the network diagram .. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.open

Re: [OpenSIPS-Users] firewall configuration

AFAIR, the default range starts from 20 000 or 30 000 .. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/20/2013 09:47 AM, Muhammad Shahzad wrote: > The default port range media proxy uses is UDP 5 - 6. You can > modify it in its c

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

Hi Qasim, Looking at the ACK related logs, I see you get the script log Sequencial 'ACK' request from caller '622190004001' for call from . twice - also the logs from the loose_route() function - I suspect you loop somehow in your script and a route is triggered twice (the route doing lo

Re: [OpenSIPS-Users] Do_Routing problem

Hello, You can check what do_routing() set (as GWs to be used) by adding this in your script (just after do_routing): xlog("Destinations are: $ru $(avp(dr_ruri)[*]) \n"); xlog("GW IDs are : $(avp(dr_gw)[*]) \n"); xlog("GW ATTRs are: $(avp(dr_gw_attrs)[*]) \n");

Re: [OpenSIPS-Users] memory consumed by t_relay

Hello , When you get the memory error, could you please do : "opensipsctl fifo get_statistics all" and send me the output (off list) please? Also, at step 4) , by stop, you mean you shutdown OpenSIPS or you stop the sipp load but OpenSIPS still runs ? Going back to my request on memory debugging

[OpenSIPS-Users] Do_Routing problem

Hi, I am using opensips 1.8.3 I set three gateways in carrier route , the call always go to the 1st and jump to third gateway and ignores the second Moreover I can see in debug level 4 , DBG:drouting:push_gw_for_usage: adding gw [gw1] in order 0 adding gw [gw2] in order 1 adding gw