[SR-Users] Try to build the SER 2.0 RC1 on Solaris 10 but failed.
I try to build the SER using gmake, after resolving some dependency issue, the script can run but finally got a error as follows. yacc -d -b cfg cfg.y conflicts: 1 shift/reduce gmake: *** [cfg.tab.c] Broken Pipe What is the problem? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] checking if calls are still active
Hi, Just want to ask what is the best way to check if calls are still active especially when we are not handling media? Thanks ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] checking if calls are still active
SST. On Monday 01 November 2010, Vic Jolin wrote: Hi, Just want to ask what is the best way to check if calls are still active especially when we are not handling media? Thanks ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] checking if calls are still active
But I heard that SST is only useful if the caller and callee supports it. is that true? On Mon, Nov 1, 2010 at 11:20 PM, Sergey Okhapkin s...@sokhapkin.dyndns.orgwrote: SST. On Monday 01 November 2010, Vic Jolin wrote: Hi, Just want to ask what is the best way to check if calls are still active especially when we are not handling media? Thanks ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] checking if calls are still active
On 11/01/2010 05:13 PM, Vic Jolin wrote: Hi, Just want to ask what is the best way to check if calls are still active especially when we are not handling media? Thanks Hello There is SST (Sip Session Timers RFC 4028), which is implemented in the SST module, but at least 1 UA must know the extension. Also I have been working on a IETF draft to use end-to-end OPTIONS ping to determine the operational status of a UA.(it works well with B2BUA, relatively well with stateful proxies). If you are interested I can talk more about the draft and how I propose using OPTIONS requests for this.(no changes on UAs are necessary) Marius. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] checking if calls are still active
That's interesting, can you tell me more about that draft? On Mon, Nov 1, 2010 at 11:18 PM, marius zbihlei marius.zbih...@1and1.rowrote: On 11/01/2010 05:13 PM, Vic Jolin wrote: Hi, Just want to ask what is the best way to check if calls are still active especially when we are not handling media? Thanks Hello There is SST (Sip Session Timers RFC 4028), which is implemented in the SST module, but at least 1 UA must know the extension. Also I have been working on a IETF draft to use end-to-end OPTIONS ping to determine the operational status of a UA.(it works well with B2BUA, relatively well with stateful proxies). If you are interested I can talk more about the draft and how I propose using OPTIONS requests for this.(no changes on UAs are necessary) Marius. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] checking if calls are still active
On 11/01/2010 05:24 PM, Vic Jolin wrote: That's interesting, can you tell me more about that draft? Hello I have attached the draft. There are 2 setups for using end-to-end OPTIONS ping 1. In Dialog... if one of the endpoints is a B2BUA, it can send at specific interval in-dialog OPTIONS ping (OPTIONS request, increasing cseq) to the other endpoint. Most well-behaved UAS will respond with a 481 if the dialog not longer exists as per RFC 3261 . The response to an OPTIONS is constructed using the standard rules for a SIP response as discussed in Section 8.2.6. The response code chosen MUST be the same that would have been chosen had the request been an INVITE. Other error conditions are on a 408 , 404 etc (as any in-dialog request is not guaranteed to succeed). This is already used some production environments. 2. Out of dialog. A proxy keeps sending OPTIONS request (no dialog involved here) to one of the UA as long as one of these 2 conditions exist: a. the dialog has not ended via a method specific mean (BYE for INVITE etc) b. a 408 (or other error codes) was received when generating a BYE. These where the big two points of the draft. Marius On Mon, Nov 1, 2010 at 11:18 PM, marius zbihlei marius.zbih...@1and1.ro mailto:marius.zbih...@1and1.ro wrote: On 11/01/2010 05:13 PM, Vic Jolin wrote: Hi, Just want to ask what is the best way to check if calls are still active especially when we are not handling media? Thanks Hello There is SST (Sip Session Timers RFC 4028), which is implemented in the SST module, but at least 1 UA must know the extension. Also I have been working on a IETF draft to use end-to-end OPTIONS ping to determine the operational status of a UA.(it works well with B2BUA, relatively well with stateful proxies). If you are interested I can talk more about the draft and how I propose using OPTIONS requests for this.(no changes on UAs are necessary) Marius. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users Network Working Group M. Zbihlei Internet-Draft 1and1 Internet AG Expires: February 2, 2011August 2010 Draft-mzbihlei-end-to-end-OPTIONS-ping-01 Abstract For VoIP providers, a common problem is related with finding the state of a dialog in certain error conditions that are caused by network problems, User Agent(UA) crashes etc. This document describes a procedure for using the Session Initiation Protocol (SIP) OPTIONS method in order to allow a SIP entity to discover the status of a UA and decide the state of the dialog. Status of this Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as work in progress. This Internet-Draft will expire on February 2, 2011. Copyright Notice Copyright (c) 2010 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Zbihlei Expires February 2, 2011[Page 1] Internet-Draft End-to-end SIP OPTIONS Ping August 2010 1. Introduction Given the multitude of parties involved in a SIP Call, a common problem for providers is determining the status of the caller/callee during a dialog. There are methods for making SIP enabled networks robust and efficient by providing redundancy at hardware or service level, but this is not the case for most UAs. Hardware problems, software crashes, power failures are factors in the way a UA behaves, affecting
[SR-Users] using tls from modules_s instead of core
Hi all. What changes i have to make in the build to use tls from modules_s instead from core modules. Thanks Jijo ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Setting up TLS between proxy and authentication server
On Oct 29, 2010 at 09:25, Jijo realj...@gmail.com wrote:
Hi Andrei,
Which version has this change? I don't see it in 3.0.4, the realease note
says that it is fixed. Here is the function
int tls_h_fix_read_conn(struct tcp_connection *c)
{
int ret;
struct tls_extra_data* tls_c;
ret = -1;=== Isn't it to be zero. Thats what i understood from the patch.
It should be 1.
The patch should be in 3.0.4 (checking git the commit is on the
kamailio_3.0 and sr_3.0 branches).
Where did you get the sources from?
Andrei
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[SR-Users] Kamalio 3.1, Siremis 2.0 Problem
I am running into the following errors: /usr/local/sbin/kamailio[6774]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Unknown column 'random' in 'field list' /usr/local/sbin/kamailio[6774]: ERROR: core [db_query.c:130]: error while submitting query /usr/local/sbin/kamailio[6774]: ERROR: sqlops [sql_api.c:217]: cannot do the query I believe it is preventing my LCR from working properly. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Dead Charts in Siremis 2.0
I managed to get the SHM chart working by manually adding the field random inside the table statistics. However, the Charts Load Charts and TM Charts gave me Internal System Errors unless I add the fields tm_active, rcv_req_diff, fwd_req_diff, and 2xx_trans_diff to the statistics table in mysql. I am not sure if these statistics are being updated, because my charts dont appear to be displaying information. Can anyone speak to what is going on here? TGF ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] New Gateway Configuration
My final question/comment for the day, I promise. I am used to using Siremis 1.x or configuring gateways. We are testing a new box on Kam 3.1 and Siremis 2.0. Previously to add an external LCR gateway, I would plop in the gateway name, group id, and tag to the LCR Gateway section and create an LCR rule with the LCR ID, Group ID, Prefix, and Priority that points to the LCR gateway that I created and I could fly straight. Since there are new attributes in 3.1 and in Siremis, how does the new configuration look? I have successfully called from softphone subscriber to softphone subscriber with no issues, but cannot get the LCR to allow me to dial beyond my internal network. Ideas? TGF ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
