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
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) {
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);
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
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
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.
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
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
___
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
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
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_
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
#
#
#
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
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
_
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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
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
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
__
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
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