[OpenSIPS-Users] Problems with SUBSCRIBES from Cisco firmware 7.4.8

Hi everyone, I've noticed that I'm having issues with successive subscribes from Cisco phones with firmware 7.4.8. Here's a SIP trace of a 2nd subscribe (when the first one is about to expire) SUBSCRIBE sip:regdev:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.3:7813;branch=z9hG4bK-9160ffcb..From:

Re: [OpenSIPS-Users] var/avp persistence, onreply_route, and script context

Hi Bobby, The $DLG_flags or a $dlg_var(name) are the pseudo variables that should help you. In order to use them, you must use create_dialog() [1] for the Initial INVITE. In the sequential requests, you should read the values only after loose_route() [2] is called, because there's where the

[OpenSIPS-Users] Multiple Presence Servers

Hi, I'd like to know if it is possible to have multiple presence servers. The idea is to distribute SUBSCRIBE messages around so that each presence server is aware of all subscriptions. However, it seems as though there is a problem with the to_tag when doing this since it is used as a matching

Re: [OpenSIPS-Users] subscribe non case sensitive user from sip uri

Hi, Ok, but also, registrar module support non case sensitive sip username. -- Dani Popa On 8/5/11 11:40 AM, Vlad Paiu wrote: Hello, What you're asking for is against the RFC 3261 URI comparison rules, which states that comparison of the userinfo part of the URI should be done case

[OpenSIPS-Users] Building an Opensips cluster with presence - help needed

Hi Anca, Bogdan, and list, I've been banging my head against this for some time now, so I'm wondering what I'm trying to do is possible. My goal is to create an Opensips cluster that provides the following: 1) Registration for phones behind NAT 2) BLF/Presence information (through Event: dialog)

[OpenSIPS-Users] OpenSIPS - true proxy mode

Hi everyone, I am trying to understand is it possible to use OpenSIPS run in full proxy mode - in other words is it possible to avoid displaying the full call information in the SIP header when sending invite packet to a voip provider to terminate the call. I have done some tests and it seems SIP

Re: [OpenSIPS-Users] OpenSIPS - true proxy mode

On Fri, Aug 5, 2011 at 2:15 PM, s...@veliko-turnovo.com wrote: is it possible to avoid displaying the full call information in the SIP header when sending invite packet to a voip provider to terminate the call. Hello, I think you are perhaps mixing up some symantics. OpenSIPs is a proxy. A