Re: [OpenSIPS-Users] reason headers

I got feedback on the ticket i opened at NEC/Philips on the question why a 400 bad packet response is sent. This is their answer: This time SV8100 responded a 400 Bad because the received message did not meet RFC. According to RFC, the text following after 'text=' should be enclosed in double quo

Re: [OpenSIPS-Users] reason headers

It's already fixed, at least in trunk. Not sure about 1.7 branch, but you can check it in the code ;-) On Jan 10, 2012, at 10:25 AM, Jan D. wrote: > I got feedback on the ticket i opened at NEC/Philips on the question why a > 400 bad packet response is sent. > > This is their answer: > > This

Re: [OpenSIPS-Users] reason headers

Hi, Thanks for the quick response. I will test vrom trunk later and will let you know if its fixed. For now, we run a production server under 1.7.1 and have problems at this moment. So I have still the question, can I get rid of the Reason header or change the text (add quotations?). -- View t

Re: [OpenSIPS-Users] reason headers

On Jan 10, 2012, at 10:52 AM, Jan D. wrote: > Hi, > > Thanks for the quick response. I will test vrom trunk later and will let you > know if its fixed. > > For now, we run a production server under 1.7.1 and have problems at this > moment. > > So I have still the question, can I get rid of the

[OpenSIPS-Users] Modify Via header

Hello, What is the best way to replace or modify Via header of incoming INVITE? I need to change private ip address with $si. Oryginal header is Via: SIP/2.0/UDP 10.10.10.128:5060;branch=z9hG4bK-680826 Is it subst? What is your advice? Regards, Maciej ___

Re: [OpenSIPS-Users] [NEW] Cassandra Database Driver

That looks very good, do you know if there is scope in the future to store SIP registration information in Cassandra (or KVP) so that a multi location registration platform can be achieved? On 5 January 2012 13:50, Vlad Paiu wrote: > ** > Hello, > > A new *CACHEDB_CASSANDRA* module has been adde

Re: [OpenSIPS-Users] [NEW] Cassandra Database Driver

Hello Pete, Yes, we are also planning to support distributed User Location, through the Key-Value Interface. After that, you will be able to use Cassandra or whatever other cachedb_* module in order to use your preferred back-end and support the functionality. So far we have implemented dist

Re: [OpenSIPS-Users] [NEW] Cassandra Database Driver

I just wanted to chime in and say that I think this direction is great. Love the distributed functionality! :) Thanks Vlad! -Brett On Tue, Jan 10, 2012 at 8:05 AM, Vlad Paiu wrote: > ** > Hello Pete, > > Yes, we are also planning to support distributed User Location, through > the Key-Value In

Re: [OpenSIPS-Users] Modify Via header

none, I think you want and need to use topology_hiding() from dialog module. Dani On Tue, Jan 10, 2012 at 1:21 PM, Maciej Bylica wrote: > Hello, > > What is the best way to replace or modify Via header of incoming INVITE? > I

[OpenSIPS-Users] Problems with DB_TEXT

I am trying to just use DB_TEXT. When I start opensips I get this error. I am using opensips 1.7 on archlinux 64bit. ERROR:db_text:dbt_query: table does not exist! ERROR:domain:reload_domain_table: Error while querying database ERROR:db_text:dbt_query: table does not exist! ERROR:drouting:dr_

[OpenSIPS-Users] OpenSIPS in a University Environment

Does anyone know of any Universities running OpenSIPS for local call routing between handsets? We're looking at replacing our old Avaya system or upgrade it, and the forklift upgrade from Avaya is ridiculously expensive (no surprise). We'd like to replace our Avaya system with a combination of Op

Re: [OpenSIPS-Users] OpenSIPS in a University Environment

Hi Gabe, Penn's serving about 9000 Polycom handsets with a combination of SERv3 (proxy), OpenSIPS (presence) and Asterisk (media) servers. We use Cisco 3845 gateways. Feel free to contact me off-list. -- Jorj -- Jorj Bauer Manager of Engineering, Research and Development Information Systems a

Re: [OpenSIPS-Users] Problems with DB_TEXT

I figured it out. I needed to escape the ":" like so \: -Original Message- From: discodog62 To: users Sent: Tue, Jan 10, 2012 5:14 am Subject: [OpenSIPS-Users] Problems with DB_TEXT I am trying to just use DB_TEXT. When I start opensips I get this error. I am using opensips 1.

[OpenSIPS-Users] one way dispatcher

Hi, Is it possible for the dispatcher module just forward the message to the selected destination (an Asterisk server) without adding a Via field to the header? In other words, I would like Asterisk to get the message as if it's coming from the original source. Thanks, Matt

Re: [OpenSIPS-Users] one way dispatcher

On Tue, Jan 10, 2012 at 2:51 PM, Matt Hamilton wrote: > > Hi, > > Is it possible for the dispatcher module just forward the message to the > selected destination (an Asterisk server) without adding a Via field to the > header? In other words, I would like Asterisk to get the message as if it's > c

Re: [OpenSIPS-Users] OpenSIPS in a University Environment

You may want to get in contact with uniza.sk. I believe they are using opensource voip as they have published some excellent tutorials like http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-and-rtpproxy-server even though it is about kamailio and not opensips On 01/10/2012

Re: [OpenSIPS-Users] one way dispatcher

>>Is it possible for the dispatcher module just forward the message to the >>selected destination (an Asterisk server) without adding a >>Via field to the header? In other words, I would like Asterisk to get the >>message as if it's coming from the original source. >Not if your are proxying th

[OpenSIPS-Users] SIP re-register time

Hi All, Could someone help me with to find what is the maximum period of time that Opensips can keep a sip-registration. I want to find out the greatest time period that I can set for phones for sending re-register requests. Regards, Yasiru___ Users

[OpenSIPS-Users] B2BUA on OpenWRT router

Hi all, I would like to setup a B2BUA SBC on our OpenWRT based router to work with our sipX PBX. Very simple requirements: authenticate outgoing calls for our ITSP preferably enforce negotiation of G729 due to limited bandwidth no need to relay RTP, just let it get passed through from the phones