Re: [OpenSIPS-Users] Opensip on Solaris

2012-08-12 Thread Nathaniel L Keeling
Thanks, I do have an issue when I perform the install. I was able to get a successful compile with a change to sipcapture.h. It was getting an error with an "unknown type name u_int32_t" so I added the definition to the file. When I performed the install, it failed due to "undefined symbol, get

Re: [OpenSIPS-Users] httpd module not working

2012-08-12 Thread duane . larson
I ran the sample OpenSIPS config that comes with all installs and I was able to start OpenSIPS with the httpd and mi_http modules. Not sure why its not starting with my config. I will send you the debug to you directly after I send this email. On , Ovidiu Sas wrote: Can you enable debug p

Re: [OpenSIPS-Users] httpd module not working

2012-08-12 Thread Ovidiu Sas
Can you enable debug probes and print out the output. Also, you can try to run a very simple config and load only httpd and mi_http modules. Regards, Ovidiu Sas On Sun, Aug 12, 2012 at 9:45 PM, wrote: > I wanted to try out the mi-http module but I am not able to get OpenSIPS to > start up when

[OpenSIPS-Users] httpd module not working

2012-08-12 Thread duane . larson
I wanted to try out the mi-http module but I am not able to get OpenSIPS to start up when I have enabled the httpd module. When I start OpenSIPS I am seeing the following error Aug 12 20:40:03 SIPProxy02 /usr/local/sbin/opensips[1537]: NOTICE:presence:child_init: init_child [-2] pid [1537]

Re: [OpenSIPS-Users] Multiple OpenSIPS instances accessing one rtpproxy?

2012-08-12 Thread bogdan
Hi Jan, Yes you can use same set of rtpproxies from multiple opensips instances. Just be aware that in this case the timeout notifications will not work (timeou events sent back to opensips). Regards, Bogdan Sent from Samsung MobileJan Blom wrote:Hello,   Would it be possible to have multi

[OpenSIPS-Users] Multiple OpenSIPS instances accessing one rtpproxy?

2012-08-12 Thread Jan Blom
Hello, Would it be possible to have multiple running instances of OpenSIPS to use a common set of RTPProxies? If not, what would break? Best regards, Jan Blom ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/

Re: [OpenSIPS-Users] Web Sockets VIA header parsing support in OpenSIPs

2012-08-12 Thread Muhammad Shahzad
Many people have already asked me on how to setup OverSIP, so here is my reply which i posted to Kamailio mailing list about a week ago, http://lists.sip-router.org/pipermail/sr-dev/2012-August/016112.html Feel free to ask if you need further help. Thank you. On Sun, Aug 12, 2012 at 5:04 PM, M

Re: [OpenSIPS-Users] Web Sockets VIA header parsing support in OpenSIPs

2012-08-12 Thread Muhammad Shahzad
This patch only allows parsing of WS VIA in OpenSIPs, you still need some WS to SIP proxy like OverSIP to receive calls from WebRTC clients. So just setup OverSIP proxy and configure it to forward all traffic from WebRTC clients to OpenSIPs, and then you can do everything that OpenSIPs can do for t

Re: [OpenSIPS-Users] Opensip on Solaris

2012-08-12 Thread Bogdan-Andrei Iancu
Hi Nathaniel, I know several people / company successfully running OpenSIPS on SPARC with Solaris - this is a supported arch / distro. By any chance, if you have issues, let me know. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 08/10/2012

Re: [OpenSIPS-Users] Dialog Custom AVP_Timeout is not valid !

2012-08-12 Thread SamyGo
Thanks Sir, It worked and really helpful tip. Good day to you. Regards, Sammu On Aug 12, 2012 1:50 PM, "Muhammad Shahzad" wrote: > First make sure you create dialog (if it is already not created by any > other module) and then set AVP before t_relay, e.g. > > create_dialog(); > $avp(10) = 180; >

Re: [OpenSIPS-Users] Web Sockets VIA header parsing support in OpenSIPs

2012-08-12 Thread SamyGo
Thats a great news. Can we get a small introduction about using this too? Regards, Sammy On Aug 12, 2012 1:49 PM, "Muhammad Shahzad" wrote: > Hi, > > Just submitted a patch to OpenSIPs developers list, which allows parsing > of WS protocol in VIA header. You can see the thread at, > > > http://s

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

2012-08-12 Thread Vlad Paiu
Hello, The <> are only required if you want to have SIP header parameters for the TO header. Otherwise, if there are no <> , all the parameters are considered to be SIP URI parameters. So, from what I see, that TO header is correct. Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-s

Re: [OpenSIPS-Users] Dialog Custom AVP_Timeout is not valid !

2012-08-12 Thread Muhammad Shahzad
First make sure you create dialog (if it is already not created by any other module) and then set AVP before t_relay, e.g. create_dialog(); $avp(10) = 180; t_relay(); exit; Thank you. On Wed, Aug 1, 2012 at 10:09 AM, SamyGo wrote: > Hi, > Im trying to implement call duration limits and I've f

Re: [OpenSIPS-Users] Opensips in the world

2012-08-12 Thread Muhammad Shahzad
Well, our SBC itself is based on OpenSIPs. It really stable and highly scale-able. For PSTN termination, e.g. T1, E1, SS7 etc. you can use either Asterisk or Freeswitch, both very good. Generally OpenSIPs is not used by small providers, its mostly used by medium sized (e.g. us) to large providers

Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

2012-08-12 Thread Bogdan-Andrei Iancu
Duane,some stupid question : are you sure your opensips is listening on the given IP:port ? have you check with netstat ? also have you checked with netstat also if there is traffic queued on the sockets ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solution

Re: [OpenSIPS-Users] Ack Ignored

2012-08-12 Thread Bogdan-Andrei Iancu
Hi Tim, That ACK is an hop-by-hop ACK (ACK to a negative reply) and it should be absorbed (with timer cancellation) by TM module when hitting any TM function (t_newtran, t_check_trans, t_relay, etc). Usually you do this in the main route (the request route) (has nothing to do with the failure

Re: [OpenSIPS-Users] Sip user behind a NAT

2012-08-12 Thread Ali Pey
Good. Have you been able to capture the call using tcpdump or ngrep? If so, do you see the Ack to 200 OK reaching the caller from callee? Meanwhile, try to use record_route_preset instead of record_route. That probably will fix it. Regards, Ali Pey On Tue, Aug 7, 2012 at 5:53 PM, Ignacio Gonza

[OpenSIPS-Users] opensips + openims

2012-08-12 Thread Kamarudin Ahmadi
hi franz how do you attached opensips.cfg and i-cscf ? best regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Web Sockets VIA header parsing support in OpenSIPs

2012-08-12 Thread Muhammad Shahzad
Hi, Just submitted a patch to OpenSIPs developers list, which allows parsing of WS protocol in VIA header. You can see the thread at, http://sourceforge.net/tracker/?func=detail&aid=3545859&group_id=232389&atid=1086412 This patch should work with any OpenSIPs version, trunk or stable 1.6.x, 1.7.