[OpenSIPS-Users] [RLS] rate of notifications and max size of NOTIFY

Hi Using RLS is there a way to control the rate of notifications and the maximum size of NOTIFY packets (partial notification) ? Our application is running on mobile phones and we don't want to receive a big NOTIFY packets if our client is subscribed to 20 or more buddies. Regards, Pascal

Re: [OpenSIPS-Users] [RLS] rate of notifications and max size of NOTIFY

Hi Pascal, No, there is no mechanism now to limit the size of the Notify, but I understand this could be useful. Please open a feature request with this and maybe we will implement it after the release. Anca Pascal Maugeri wrote: Hi Using RLS is there a way to control the rate of

Re: [OpenSIPS-Users] Opensips as SBC

Hi Ghaith, As Brett says, all these functionalities can be achieved via OpenSIPS - there are more than 100 module offering functionalities you may need - routing, traffic shaping, QoS, etc... Maybe you should ask, for each functionality you need, a more precise question so that somebody can

Re: [OpenSIPS-Users] Pass calls to another realm via gateway

Hello Alexander, For the initial request (like INVITE), you can set a simple logic on opensips: 1) if does not come from platform - forward to the platform 2) if does come from platform - route based on RURI (this will cover the foreign domains also) For sequential requests (ACK, BYE,

Re: [OpenSIPS-Users] Re-invite problem - 491 Request Pending

Hi Brett, This is an ancient topic that needs to be solved once for all. The bottom problem is that OpenSIPS / TM does try o match the 200OK ACK against the INVITE transaction - and it should not do that as 200OK ACK forms a separate transaction and it matches at dialog level, not transaction

Re: [OpenSIPS-Users] [RLS] rate of notifications and max size of NOTIFY

I opened one artifact for this purpose: http://sourceforge.net/tracker/?func=detailaid=2873450group_id=232389atid=1086413 Regards, Pascal On Tue, Oct 6, 2009 at 2:34 PM, Anca Vamanu a...@opensips.org wrote: Hi Pascal, No, there is no mechanism now to limit the size of the Notify, but I

Re: [OpenSIPS-Users] [RLS] rate of notifications and max size of NOTIFY

Hi, On Tue, Oct 6, 2009 at 3:01 PM, Pascal Maugeri pascal.maug...@gmail.com wrote: I opened one artifact for this purpose: http://sourceforge.net/tracker/?func=detailaid=2873450group_id=232389atid=1086413 Does OpenSIPS support RFC4483? Knowing the size in advance, a determination about whether

Re: [OpenSIPS-Users] Re-invite problem - 491 Request Pending

Bogdan,I presently record the 200 OK ACK in my ACC, but I don't seem to actually utilize it for anything at present. If I did the fix jeff mentioned, will I no longer get that ACK in ACC? so performing the t_check_trans() is faster than tm module's built in matching? I'm not sure I get why this

[OpenSIPS-Users] 17 sec, recieve a bye and a hangup

I'm trying to intergrate opensips with a allready running Asterisk server. The two servers are both on the same machine. I can recieve calls fine, Asterisk send them to my opensips installation, and the opensips forwards the phone call to the right user. I can call between the users on the

Re: [OpenSIPS-Users] 17 sec, recieve a bye and a hangup

I guess the question here is, what is asterisk doing for you? I personally would prefer the sip trunks right on opensips.. Asterisk is a kinda funny bottleneck in your architecture unless it's acting as some sort of media server (or TDM gateway). Some potential issues: 1. Do you have 2 way audio,

Re: [OpenSIPS-Users] 17 sec, recieve a bye and a hangup

I understand you can find it under this text. as you can see, the call just disapeare, i see now that the bye appears when i hang up the polycom phone. I hope this information helps. U 172.16.0.12:5060 - 172.16.1.10:5090 INVITE sip:0624469...@172.16.1.10:5090;user=phone SIP/2.0. Via:

Re: [OpenSIPS-Users] 17 sec, recieve a bye and a hangup

Ok, so as you can see from the trace, the 200OK is retransmitted.. you'll see:200OK ACK 200OK 200OK 200OK 200OK BYE This is because the ACK never made it to Astersk. It's a scripting error in Opensips. You may want to check your loose route block for errors. Also, it's worth mentioning that the

Re: [OpenSIPS-Users] Re-invite problem - 491 Request Pending

Bogdan, I apologize for 'beating a dead horse'. I get that this is a frustration we're stuck with for various reasons. I've been writing my Opensips config on a daily basis for going on six months now, and there's still a couple of weird spots in my scripts that drive me nuts (I

[OpenSIPS-Users] Login to OpenSIPS website

Hi, I have been trying to login to OpenSIPS website using the Login option. However, everytime I click on the confirmation email link, I see a webpage with error messages. From existing email account. Error: Your email link is not synchronized with the state of the password file. Please

[OpenSIPS-Users] Prepay charge by source IP

I am trying to setup a opensips/cdrtool/callcontrol gateway for pre-pay ratings. My conundrum is that I need to debit account by IP alone (not username). I found that I can auth by IP by just inserting it into the database, but cannot seem to wild-card the username. Is there a way to do this

[OpenSIPS-Users] Has anyone implemented account codes?

We'd like to implement the ability to organize CDRs by account code, and/or to restrict calling to certain areas based on code. Rather than delve into this in detail, I am linking Verizon's explanation of the service: http://www22.verizon.com/longdistance/pdf/plan_acctcodes_east.pdf Any input