Re: [OpenSER-Users] Routing Logic

2008-05-13 Thread VoIP Forums www . Go4Calls . com
Hi all, Thank you very much for the explainion it is working well now. and thank you very much Edson to providing me the right code and it worked exact. Regards, www.Go4Calls.Com VoIP Forums > Date: Tue, 13 May 2008 15:55:10 +0200 > From: [EMAIL PROTECTED] > To: users@lists.openser.org > Su

[OpenSER-Users] Routing Logic

2008-05-13 Thread VoIP Forums www . Go4Calls . com
Hi Friends, I have the following route plan in my openser.cfg all blocks are working well instead of 800800. if (uri=~"sip:[EMAIL PROTECTED]") { rewritehostport("officePBX-IP:5060"); route(1); exit; } else if ($(rU{s.len})>=8) {

[OpenSER-Users] URI Length

2008-03-27 Thread VoIP Forums www . Go4Calls . com
Hi All, How i can get the length of called number. i am trying to making routing logic base on length. Please help us how i can get the length i tried the fillowing but did not work. if (uri:len<=7) { rewritehostport("192.168.1.1:5060"); route(1);

Re: [OpenSER-Users] Local Call

2008-02-07 Thread VoIP Forums www . Go4Calls . com
sts.openser.org > > Use curly brackets after each "else if" line. > > > Regards, > Ovidiu Sas > > On Feb 7, 2008 5:12 AM, VoIP Forums www. Go4Calls. com > <[EMAIL PROTECTED]> wrote: > > > > Hi friends, > > > > Could you please he

[OpenSER-Users] Local Call

2008-02-07 Thread VoIP Forums www . Go4Calls . com
Hi friends, Could you please help me how i can make dialplan to call local each subscriber using some special prefix? I tried the following but it did not work. if (uri=~"sip:[EMAIL PROTECTED]") { strip(2); rewritehostport("pstncarrier:5060"); ro

[OpenSER-Users] Call Session Time

2008-02-06 Thread VoIP Forums www . Go4Calls . com
Hi all, How i can store Call Session Time in acc table? I am using Mediaproxy so i hope the call duration will be correct event if openser does not receive the BYE. I tried to use ACC but inside the table all information is storing only i cannot see the filed name for session or call duration.

Re: [OpenSER-Users] Accurate CDR with RTPProxy

2008-02-05 Thread VoIP Forums www . Go4Calls . com
HI, Can we install Mediaproxy on NAT mode server? and Openser in NO-NAT? after that openser will communicate Mediproxy server using LAN IP? Regards, www.Go4Calls.Com VoIP Forums > Date: Tue, 5 Feb 2008 11:47:00 +0100 > From: [EMAIL PROTECTED] > To: users@lists.openser.org > Subject: Re: [O

[OpenSER-Users] Accurate CDR with RTPProxy

2008-02-05 Thread VoIP Forums www . Go4Calls . com
Hi all, Is it possible to get accurate CDR of openser using RTPProxy? If ATA did not send BYE message to openser can RTPProxy disconnect the call using some rtptimeout then ACCT module log the correct duration in MySQL database? Regards, www.Go4Calls.Com VoIP Forums ___

Re: [OpenSER-Users] Calls disconnect automatically

2008-01-21 Thread VoIP Forums www . Go4Calls . com
Hi All, I have captured the packet when the device sent ICMP to the media gateway and the reply comming "DESTINATION PORT UNREACHABE" that time device send BYE message to Openser. I am not so expert to analysis the packet if you could have a look to the following URL i put already the shot of

Re: [OpenSER-Users] Calls disconnect automatically

2008-01-20 Thread VoIP Forums www . Go4Calls . com
The problem is only with PSTN call. I tried to send call to the three gateway Teles, SIP-HIT and Asterisk but all disconnect calls in that priticular seconds. The thinng is i cannot understand if i am using STUN in Linksyspap2 the call goes normal and without STUN it disconnect. So the problem

Re: [OpenSER-Users] Calls disconnect automatically

2008-01-20 Thread VoIP Forums www . Go4Calls . com
Hi, i tired with the following configuration but still result is same. calls disconnect in 30 - 32 sec modparam("nathelper", "natping_interval", 20) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_

Re: [OpenSER-Users] Calls disconnect automatically

2008-01-20 Thread VoIP Forums www . Go4Calls . com
Hi sorry, I forget to give my openser.cfg, there is one more point if i am using STUN server in our linksys device the call goes normal for long time till user finish the call. It seems something wrong in NAT configuration. # # sample config file to be used with nathelper/rtpproxy # # #

[OpenSER-Users] Calls disconnect automatically

2008-01-20 Thread VoIP Forums www . Go4Calls . com
Hi Friends, I start getting one problem, the calls disconnect automatically in 30 and 32 sec. I am using openser + rtpproxy before with the same openser.cfg it was running smoothly and once traffic increased this problem appeared. Could you please help me to solve this issue because i put open

[OpenSER-Users] Saving Active Calls

2007-12-22 Thread VoIP Forums www . Go4Calls . com
Hi Friends, How we can save Active Calls in openser mysql database? Please advise us i need to display it on our billing system. Regards, www.Go4Calls.Com VoIP Forums _ Express yourself instantly with MSN Messenger! Download to

Re: [OpenSER-Users] Display Registered Users

2007-12-12 Thread VoIP Forums www . Go4Calls . com
Hello bogdan, Thank you very much for your help. Now i am able to get registered users in detial and also in short. Regards, www.Go4Calls.Com VoIP Forums > Date: Wed, 12 Dec 2007 12:38:35 +0200 > From: [EMAIL PROTECTED] > To: [EMAIL PROTECTED] > CC: [EMAIL PROTECTED] > Subject: Re: [OpenSE

[OpenSER-Users] Display Registered Users

2007-12-12 Thread VoIP Forums www . Go4Calls . com
Hi All, Could you please guide us with right cmd to display the users which are registered successfully with openser? Regards, www.Go4Calls.Com VoIP Forums _ Express yourself instantly with MSN Messenger! Download today it's FR

[OpenSER-Users] Set CallerID as Username

2007-12-11 Thread VoIP Forums www . Go4Calls . com
Hi All, i was routing calls from openser to asterisk to make accounting in asterisk side. I need small help, how i can set callerid = registrar username so in asterisk server i can capture the callerid to do the billing etc Thank You __

[OpenSER-Users] Calls Between Registered Accounts

2007-12-01 Thread VoIP Forums www . Go4Calls . com
Good day friends, I was trying to install Openser and Asterisk on same server. I need some configuration or information if anyone can point me. I need to forward only International call to the Asterisk but the same time if registered accounts of openser want to make call between them the can c