Re: [OpenSIPS-Users] RLS change from the latest trunk svn

Hi Anca, I tried with the xml document that you have mentioned in the below mail.. rls_handle_subscribe is failing. Below is the log: Nov 14 14:48:48 info opensips[11679]: DBG:rls:get_resource_list: rls_services document: http://www.w3.org/2001/XMLSchema-instance";> presence Nov 14 14:48:48 i

Re: [OpenSIPS-Users] how can opensips support both IPv4 and IPv6?

Hi Bogdan , I have my server sip integrated with asterisk for voicemail and meetme, adds to rtpproxy to solve my problems of nat with my remote clients, the result was almost satisfactory some details to improve, the problem was that after adding rtpproxy when my clients do not answer a call no lon

Re: [OpenSIPS-Users] Drop call if mediaproxy does not work (using engage_mediaproxy())

On Thursday 13 November 2008, Giuseppe Roberti wrote: > Dan Pascu wrote: > > On Thursday 13 November 2008, Giuseppe Roberti wrote: > >> Hi. > >> > >> How can we drop a call (an INVITE) if we are using > >> engage_media_proxy() > > > > You can't. engage_media_proxy() does nothing when called except

[OpenSIPS-Users] problem using scripting variables AND inject random delays

Hi all, i am quite new to use opensips, i have been using SER since last week. i want to use opensips to inject a delay then relay SIP messages, actually there is function "sleep" under cfgutils.so module, i have tested it .. it is working fine, my problems are: 1/ i have problem usin

[OpenSIPS-Users] drounting

Hello, I've been testing drouting, and as always with "new" modules, it works great. I've got a question: dr_groups has a "username" and "domain" field, it is my understanding that if I leave username blank and put something on domain, everything directed to that domain will be routed using t

Re: [OpenSIPS-Users] [Fwd: [Serdev] the sip router project]

YOU THE MAN I KNEW YOU'D GET THAT OUT! hehe! D On Thu, Nov 6, 2008 at 5:06 PM, Bogdan-Andrei Iancu <[EMAIL PROTECTED]>wrote: > Well, if you need more in that area, then stay tune :) > we already have the automatic call ending (from proxy) on timeout ;) > > Regards, > Bogdan > > Brett Nemerof

Re: [OpenSIPS-Users] Drop call if mediaproxy does not work (using engage_mediaproxy())

Dan Pascu wrote: > On Thursday 13 November 2008, Giuseppe Roberti wrote: >> Hi. >> >> How can we drop a call (an INVITE) if we are using engage_media_proxy() > > You can't. engage_media_proxy() does nothing when called except to set an > internal flag, so it can't return an error code because not

Re: [OpenSIPS-Users] NAT: Why replacing "Contact " with thereceivedpublic IP:port instead of adding a parameter with it?

El Jueves, 13 de Noviembre de 2008, Bogdan-Andrei Iancu escribió: > It might be right (from semantic pov) - I suggested this as there are > other devices using it for similar purpose, so there is a kind of > unwritten convention and the probability to work (to be properly > interpreted by other dev

Re: [OpenSIPS-Users] Drop call if mediaproxy does not work (using engage_mediaproxy())

On Thursday 13 November 2008, Giuseppe Roberti wrote: > Hi. > > How can we drop a call (an INVITE) if we are using engage_media_proxy() You can't. engage_media_proxy() does nothing when called except to set an internal flag, so it can't return an error code because nothing can fail in that opera

Re: [OpenSIPS-Users] Error relaying ACK to the same IP but different port

You're right Adrian, I've pasted my confing here( http://pastebin.com/m7ce61f09) It's large, and it has not "friendly" comments, but may be you can see if there are logics error. Be free to ask if you need more explanation about this issue. Thank you Gustavo On Thu,

Re: [OpenSIPS-Users] reply with CANCEL message

2008/11/13 Alex R. S. M <[EMAIL PROTECTED]>: > No CANCEL message is being sent to openSIPS. OpenSIPS received 180 from > both End-points. One End-point answers. > I want OpenSIPS to generate CANCEL message and sends to the other End-point. Ok. When OpenSIPS receives the 200 from one branch it wil

Re: [OpenSIPS-Users] reply with CANCEL message

No CANCEL message is being sent to openSIPS. OpenSIPS received 180 from both End-points. One End-point answers. I want OpenSIPS to generate CANCEL message and sends to the other End-point. > Date: Thu, 13 Nov 2008 17:45:50 +0100 > From: [EMAIL PROTECTED] > CC: users@lists.opensips.org > Subjec

Re: [OpenSIPS-Users] Error relaying ACK to the same IP but different port

Gustavo, If you post your configuration you score more chances to get useful help from somebody. What you have just showed can have an infinite number of reasons without a configuration. Adrian On Nov 13, 2008, at 6:02 PM, Gustavo Mistrinelli wrote: Hello, I have a schema composed by Open

[OpenSIPS-Users] Drop call if mediaproxy does not work (using engage_mediaproxy())

Hi. How can we drop a call (an INVITE) if we are using engage_media_proxy() but dispatcher does not work (like when not started ;) ? With use_media_proxy() we can check the return code (i hope) but if we use engage_media_proxy() this is not possible. Regards. -- Giuseppe Roberti <[EMAIL PROTECT

[OpenSIPS-Users] Error relaying ACK to the same IP but different port

Hello, I have a schema composed by OpenSIPS (10.20.200.1:*5060*) and B2BUA-Asterisk (10.20.200.1:*5070*) , Here goes a call flow from a hardphone (10.10.115.113) to a Conference Number located in another server (Asterisk: 192.168.0.1) I found an issue, see timeframe 0.028, you can see OpenSIPS sen

Re: [OpenSIPS-Users] reply with CANCEL message

2008/11/13 Alex R. S. M <[EMAIL PROTECTED]>: > End-point B sends a REGISTER message to openSIP. I get the URI of the > End-point B from "location" table using a perl script. Then I manually > generate a branch with append_branch(). It doesn't matter how you decide the new URI of the new branch. Yo

Re: [OpenSIPS-Users] reply with CANCEL message

End-point B sends a REGISTER message to openSIP. I get the URI of the End-point B from "location" table using a perl script. Then I manually generate a branch with append_branch(). Is the registrar's forking mechanism different? > Date: Thu, 13 Nov 2008 11:31:23 -0500 > From: [EMAIL PROTECT

Re: [OpenSIPS-Users] reply with CANCEL message

2008/11/13 Alex R. S. M <[EMAIL PROTECTED]>: > The INVITE request to End-point B generated with append_branch() within > openSIP. > So how openSIP knows to generate a CANCEL message when one End-point answers > the call? What is the problem? when you generate a new branch with "append_branch()" Op

Re: [OpenSIPS-Users] reply with CANCEL message

Alex R.S.M wrote: > The INVITE request to End-point B generated with append_branch() within > openSIP. > So how openSIP knows to generate a CANCEL message when one End-point > answers the call? Are you generating it manually or using the registrar's forking mechanism? -- Alex Balashov Evariste

Re: [OpenSIPS-Users] reply with CANCEL message

The INVITE request to End-point B generated with append_branch() within openSIP. So how openSIP knows to generate a CANCEL message when one End-point answers the call? Date: Thu, 13 Nov 2008 10:19:08 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: users@lists.opensips.org Subject: Re

Re: [OpenSIPS-Users] reply with CANCEL message

The cancel should go out automatically. You don't need to generate it. On Thu, Nov 13, 2008 at 10:13 AM, Robert R <[EMAIL PROTECTED]> wrote: > Hi, > > - openSIP received two 180 messages from both End points A and B. > - End-point A answers, and sends a 200OK to openSIP > Then, I need to sends a

[OpenSIPS-Users] reply with CANCEL message

Hi, - openSIP received two 180 messages from both End points A and B. - End-point A answers, and sends a 200OK to openSIP Then, I need to sends a CANCEL message to End-point B. Which one of the functions I can use: sl_send_reply( ) or t_reply( ) ? I assume the function should be called in onrep

Re: [OpenSIPS-Users] Input on my loadbalancer configuration

Hi Geoff, OpenSIPs is by default only transaction stateful, so it does matter how many concurrent calls you have. What matter is the setup and tire-down capacity (like how many calls per second can be established and terminated). In pure relay mode, you can go up to 10K calls per second (as se

Re: [OpenSIPS-Users] Perl dialog callback

Will anyone be taking over the perl module? On Thu, Nov 13, 2008 at 7:09 AM, Bogdan-Andrei Iancu <[EMAIL PROTECTED] > wrote: > > >> so, will you not be available to > >> maintain the perl module anymore ? > >> > > > > I will try to provide fixes for bugs that may occur, but I will not be > able t

Re: [OpenSIPS-Users] RLS change from the latest trunk svn

Hi Jeon, You spotted the changes very well :) . They are listed in commit log for revision *4854:* /- fixed bugs in RLS discovered at SIPIT - use rls-services document to take the list instead of resource-list

Re: [OpenSIPS-Users] CDRTool problem

Hi Dilip, Did you read the RATING.txt document? http://cdrtool.ag-projects.com/browser/doc/RATING.txt Adrian On Nov 13, 2008, at 2:27 PM, Dilip wrote: Hello Evrybody, I am using CDRTool for the accounting.It GUI is working fine. But i wants to add the rating in its GUI.For that i have read

[OpenSIPS-Users] CDRTool problem

Hello Evrybody, I am using CDRTool for the accounting.It GUI is working fine. But i wants to add the rating in its GUI.For that i have read the Rating Engine Documentation but its not affected. Whats the exact procedure to display the call rate ?? which r the tables require to edit ??? Can anybod

Re: [OpenSIPS-Users] Perl dialog callback

Ok, thanks for clarification Bastian. Regards, Bogdan Bastian Friedrich wrote: > On Thursday 13 November 2008, Bogdan-Andrei Iancu wrote: > >> Thank you for the feedback - so, if I understand correctly, your >> company is not into VOIP anymore ? >> > > Unfortunately: correct. > > >> s

Re: [OpenSIPS-Users] Perl dialog callback

On Thursday 13 November 2008, Bogdan-Andrei Iancu wrote: > Thank you for the feedback - so, if I understand correctly, your > company is not into VOIP anymore ? Unfortunately: correct. > so, will you not be available to > maintain the perl module anymore ? I will try to provide fixes for bugs

Re: [OpenSIPS-Users] Input on my loadbalancer configuration

Bogdan (and list), Again, thank you very much for taking the time to answer my questions. It is people like you that enable the open source community to thrive and grow. I really appreciate your efforts. I have gone through and made your suggested changes and things seem to be working quit

Re: [OpenSIPS-Users] DNS SRV Crashes

Hi Brett. I apologize for a error in my last mail. You would not update to 1.4.3, but you would update to SVN latest stable version 1.4 Best regards. Sergio. On 11/11/08, Sergio Gutierrez <[EMAIL PROTECTED]> wrote: > > Hi Brett. > > Is it difficult to you update to 1.4.3? > > As I can see comp

Re: [OpenSIPS-Users] Perl dialog callback

Hi, On Tuesday 11 November 2008, Bogdan-Andrei Iancu wrote: > Giuseppe Roberti wrote: > > > > Is it possible to register a user perl function as a dialog callback ? > > If so, how to do it ? > > well, this is not available right now - I'm not too much into the perl > module (how it works), but pro

Re: [OpenSIPS-Users] Perl dialog callback

Hi Bastian, Thank you for the feedback - so, if I understand correctly, your company is not into VOIP anymore ? so, will you not be available to maintain the perl module anymore ? Regards, Bogdan Bastian Friedrich wrote: > Hi, > > On Tuesday 11 November 2008, Bogdan-Andrei Iancu wrote: > >

Re: [OpenSIPS-Users] NAT: Why replacing "Contact" with thereceivedpublic IP:port instead of adding a parameter with it?

Hi Klaus, It might be right (from semantic pov) - I suggested this as there are other devices using it for similar purpose, so there is a kind of unwritten convention and the probability to work (to be properly interpreted by other devices) is higher. BTW, did you know if the RFC recommends/sp

Re: [OpenSIPS-Users] how can opensips support both IPv4 and IPv6?

See the help from rtpproxy: usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] [-t tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles] and the manual for nathelper (flags for force_rtp_proxy): http://www.opensips.org/html/docs/modules/1.4.x/nathelper.html#id2515879 Regards

Re: [OpenSIPS-Users] radius fields

The syntax is correct - I just tested it with no problem. Be careful with the var() as these variables are inherited in a process (by all routes executed by a process) and not shared across the transaction. See http://www.opensips.org/index.php?n=Resources.DocsCoreVar In what route are you

Re: [OpenSIPS-Users] Input on my loadbalancer configuration

Hi Geoff, [EMAIL PROTECTED] wrote: > Thank you very much for taking the time to look over my configuration. > I just want to make sure of something. I replied to my own original > message with a greatly enhanced configuration. I realized the first > was missing a huge amount of logic after stud

Re: [OpenSIPS-Users] Input on my loadbalancer configuration

Use t_reply() instead of sl_send_reply() and ds_select_domain() instead of ds_select_dst(): ds_select_domain("1","4"); t_reply("301","redirected"); Let me know if this works. Regards, Bogdan [EMAIL PROTECTED] wrote: > I invoked the dispatcher, but it doesn't appear anything was set. My > fl

Re: [OpenSIPS-Users] Problem with force send socket.

Hi Ola, If the VIA is set to the forced outbound interface, are you sure there is no routing rules on your system to force outbound messages from .106 to .101 ? Regards, Bogdan Ola Karlsson wrote: > Lightning fast Bogdan.. ;-) > > I think so. > #opensips > Listening on > udp: 127.