[OpenSIPS-Users] how to make a phone call to between two different domain sip server

[Michael Leung]

Hi all

i know this is a stupid question

but i dont use sip to make a phone call very often ,

i have setup up two opensips server in my intranet environment

i use two phones to register on each server

how to make a phone call from one to another one

do i have to add the the destination domain name behind the alias number
when i dial out ?

or why can i dial the alias number without domain name , then the opensips
server will routing it to a the opensips server automatically


thanks

Michael
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Re: [OpenSIPS-Users] Load balancer setup

[Matt Broad]

Hi Kenny,

thanks for the reply.  For some reason this is only just showing in my
email so sorry for the delayed response.

Could you explain how you would do this is asterisk please? This may help
in me solving the issue :)



thanks
Matt

On 23 October 2014 21:40, Kenny Watson  wrote:

>  Hi Matt,
>
> The SDP is being generated by freeswitch and you would need to make it
> think that when its sending to kamailio, that kamailio  is an external host
> so freeswitch uses its public IP address in the SDP  that it is then
> forwarded on directly to the carrier.
>
>  I've never used freeswitch but thats roughly what you'd do with asterisk.
>
> Thanks
> Kenny Watson
>
>
>   --
> *From:* users-boun...@lists.opensips.org [users-boun...@lists.opensips.org]
> on behalf of matt [m...@supportedbusiness.com]
> *Sent:* 22 October 2014 08:42
> *To:* users@lists.opensips.org
> *Subject:* [OpenSIPS-Users] Load balancer setup
>
>   Hi,
>
>
>  I was looking for some guidance on using the load balancer in a NAT
> environment.
>
>  I have the following setup (the IP addresses are made up but should give
> an indication):
>
>  1 x opensips server with load balancer module - IP 192.168.0.1
>  2 x freeswitch servers - IP 192.168.0.2 & 192.168.0.3
>
>  All 3 servers have seperate external IP address routing to their
> internal IP via our firewall:
> 217.0.0.1 routed to 192.168.0.1 (Opensips)
> 217.0.0.2 routed to 192.168.0.2 (FS1)
>  217.0.0.3 routed to 192.168.0.3 (FS2)
>
>  I have the load_balancer table with the following details:
>
>  id,  | group_id, |  dst_uri,| resources,  |
> probe_mode, | description
> '1',  |  '1', |  'sip:192.168.0.2:5080',  |   'pstn=10', |
>'2',   |  'FS1'
> '2',  |  '1', |  'sip:192.168.0.3:5080',  |   'vm=1', |
>   '2',   |  'FS2'
>
>
>  The call flow is:
>
>  SIP Provider --> 217.0.0.1 Opensips --> 192.168.0.2/3
>
>  The issue is, that when the 200 ok response is sent to the SIP provider,
> the Freeswitch server's internal IP is being sent in the SDP connection
> information (c).  This causes the ACK response from the SIP Provider to
> fail to be sent correctly.
>
>  With the calls routed directly to the FS servers (removing opensips from
> the flow), the calls work fine.
>
>  Any help would be much appreciated :)
>
>
>  thanks
> Matt
>
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Re: [OpenSIPS-Users] Content-Type Paramter issue

[chow]

I read  code in msilo module . 
It seems not contain parameter in content-type header when insert to DB,
why?
what consider for that. 
thank you any advice.!



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Re: [OpenSIPS-Users] Problem with CDRTool and ubuntu 14.04 and apache 2.4.7 and PHP 5.5

[Tijmen de Mes]

Hi,

We did not test it on 5.5 yet. The latest deb is compatible with 5.3. The code 
in darcs should work with 5.4. 

-- 
Tijmen de Mes
AG-Projects

From: Venkatesh Macha 
Reply: OpenSIPS users mailling list >
Date: 30 oktober 2014 at 11:45:44
To: users@lists.opensips.org >
Subject:  Re: [OpenSIPS-Users] Problem with CDRTool and ubuntu 14.04 and apache 
2.4.7 and PHP 5.5  

I think problem with *PHP*. I am using php 5.5, Is CDRTool is compatible  
with PHP 5.5 ??  

I saw one post on this topic it was few months back one you can see here ->  
http://lists.opensips.org/pipermail/users/2013-February/024453.html  
  

If i use php 5.3 it is working fine on ubuntu 12.04 but it is not working  
with ubuntu* 14.04 and php 5.5.*  

is CDRTool is Compatible with PHP 5.5 ???  

Venkatesh macha,  
Junior VOIP Engineer,  
@ C programming   



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Re: [OpenSIPS-Users] Mangle SDP Connection IP

[Sunil P]

Hi All,

   This is the sample mangler.cfg
https://github.com/lemenkov/sip-router/blob/master/modules_s/mangler/mangler.cfg.Now
I
need to use it for 200 ok of INVITE.Please help if anybody knows.

Thanks and Regards,
Sunil

On Fri, Oct 31, 2014 at 12:19 PM, Sunil P  wrote:

> Hi All,
>
>
>  using  sdp_mangle_ip we can change the sdp connection ip.In
> mangaler.cfg there is check if (method == "INVITE"){sdp_mangle_ip } call
> this function.Now how to call sdp_mangle_ip for 200 ok of
> INVITE.
>
> Thanks and Regards,
> Sunil
>
>
>
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