sr-users

sr-users@lists.kamailio.org

3 participants
37455 discussions

[Users] several presence question
by Klaus Darilion 03 Jan '07

03 Jan '07

Hi! I read the docs of the presence module and have some question. Maybe someone can enlighten me ;-) 1. If I understand correctly, the pua module acts as a "UAC". Thus, it can be used to send PUBLISH or SUBSCRIBE. I wonder what is the usecase for sending SUBSCRIBE? Usually, a client which sends a SUBSCRIBE receives subsequent NOTIFY messages. How can these NOTIFY messages be handled? 2. If an external application sends PUBLISH via the uac_mi interface, how can be distinguished between a "… [View More]

3 9

RE: [Users] SER as proxy in front of Asterisk
by Thomas Deillon 03 Jan '07

03 Jan '07

Hi, In fact, now, every phone does not response. The options didn't go thought the FW, so, the phone cannot answer. I don't understand why the traffic don't go thought it !! I have test this configuration with two firewall: a sonicwall and a soho. Cheers, Thomas -----Message d'origine----- De : Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Envoyé : mercredi, 3. janvier 2007 14:52 À : Thomas Deillon Cc : users(a)openser.org Objet : Re: [Users] SER as … [View More]proxy in front of Asterisk Hello, what is the type of NAT/Firewall you have in front of Thomson? As I can see from the diagram, the phone does not reply to OPTIONS request, and I guess, the nat/firewall needs outward traffic to keep the pinhole open. Check why thomson ignores the options requests. Cheers, Daniel On 01/03/07 15:17, Thomas Deillon wrote: > > Hi, > > I have made more tries and now, all my phones don't work ... > > The firewall opens a connection to REGISTER phone but after a while > the OPTION message from the Asterisk will be DROP by the FW ... > > You can find a diagram of the exchange here: > http://deillon.eu/tmp/exchange_sip.png > > A list of all exchange: http://deillon.eu/tmp/ip_sip.txt > > And all sip message: http://deillon.eu/tmp/message.txt > > Thanks a lot for your help, > > Thomas > > ------------------------------------------------------------------------ > > *De :* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *De la part de* Thomas Deillon > *Envoyé :* mardi, 2. janvier 2007 11:44 > *À :* users(a)openser.org > *Objet :* [Users] SER as proxy in front of Asterisk > > Hi all, > > I wish you first a happy new year !! > One again, I ask you some help. Thanks a lot for your patience and > your answers that really helped me. > > So, I want to put a SER in front of one Asterisk for the moment (more > after). > All Servers have public IP address and so, I don't care about the > RTP/SDP messages. > All phones are behind NAT somewhere in Switerland :) > > > I have the configuration below in openser: > > > modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") > #modparam("dispatcher", "force_dst", 1) > modparam("dispatcher", "flags", 2) > > modparam("usrloc", "db_mode", 0) > #modparam("rr", "enable_full_lr", 1) > > route{ > > > xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > }; > > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > # if (search("User-Agent:.*Thomson.*")) { > # }; > > > if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { > if(method=="OPTIONS") { > }; > avp_pushto("$ru","$tu"); > forward(); > > }else { > fix_nated_contact(); > force_rport(); > if(method=="REGISTER"){ > ds_select_dst("4", "0"); > t_relay(); > } > else { > ds_select_dst("0", "4"); > forward(); > } > } > } > > > And in my dispatcher.list I have: > > 0 sip:212.xxx.xxx.153:5060 > 4 sip:212.xxx.xxx.153:5060 > > > > With this configuration, a snom phone, a X-lite phone and a Cisco > phone seams to work .... > But with the Thomson ST2030, it doesn't work. > In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION > message to the FW IP adress and the right port where the thomson is > and then, after a while, the FW close the connection. The thomson > phone is so "UNREACHABLE" on asterisk status and nobody can call it. > I'm not sure that my configuration is ok in ser but I think that is a > problem with the Thomson2030. Do you had the same kind of problem or > do you understand the problem here ? > > > Thanks for your help, > > Thomas Deillon > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > [View Less]

1 0

RE: [Users] SER as proxy in front of Asterisk
by Thomas Deillon 03 Jan '07

03 Jan '07

Hi, I have made more tries and now, all my phones don't work ... The firewall opens a connection to REGISTER phone but after a while the OPTION message from the Asterisk will be DROP by the FW ... You can find a diagram of the exchange here: http://deillon.eu/tmp/exchange_sip.png A list of all exchange: http://deillon.eu/tmp/ip_sip.txt And all sip message: http://deillon.eu/tmp/message.txt Thanks a lot for your help, Thomas ________________________________ De : … [View More]users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Thomas Deillon Envoyé : mardi, 2. janvier 2007 11:44 À : users(a)openser.org Objet : [Users] SER as proxy in front of Asterisk Hi all, I wish you first a happy new year !! One again, I ask you some help. Thanks a lot for your patience and your answers that really helped me. So, I want to put a SER in front of one Asterisk for the moment (more after). All Servers have public IP address and so, I don't care about the RTP/SDP messages. All phones are behind NAT somewhere in Switerland :) I have the configuration below in openser: modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") #modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 2) modparam("usrloc", "db_mode", 0) #modparam("rr", "enable_full_lr", 1) route{ xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # if (search("User-Agent:.*Thomson.*")) { # }; if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { if(method=="OPTIONS") { }; avp_pushto("$ru","$tu"); forward(); }else { fix_nated_contact(); force_rport(); if(method=="REGISTER"){ ds_select_dst("4", "0"); t_relay(); } else { ds_select_dst("0", "4"); forward(); } } } And in my dispatcher.list I have: 0 sip:212.xxx.xxx.153:5060 4 sip:212.xxx.xxx.153:5060 With this configuration, a snom phone, a X-lite phone and a Cisco phone seams to work .... But with the Thomson ST2030, it doesn't work. In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION message to the FW IP adress and the right port where the thomson is and then, after a while, the FW close the connection. The thomson phone is so "UNREACHABLE" on asterisk status and nobody can call it. I'm not sure that my configuration is ok in ser but I think that is a problem with the Thomson2030. Do you had the same kind of problem or do you understand the problem here ? Thanks for your help, Thomas Deillon [View Less]

2 1

[Users] 1.1.0 Compiling for Solaris 10
by Martin Klisch 03 Jan '07

03 Jan '07

Hi, i can not compile openser 1.1.0 for solaris 10. any idea why? gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.3"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -… [View More]

1 1

[Users] Problem registering the UA with openSER(tls enabled)
by Ncheeku Baranov 03 Jan '07

03 Jan '07

Hi, I am trying to make my non-TLS/TLS UA register with my TLS enabled openSER. Currently I am just working on my local machine with the client UAs on the same subnet,(so there is only one domain, but its not named). Below is my configuration file: disable_tls = 0 listen = tls:10.30.100.41:5061 tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user- cert.pem" tls_private_key = "/usr/local/etc/… [View More]

3 11

[Users] 487 Transaction Terminated
by KOK FOON YEW 03 Jan '07

03 Jan '07

This is pertaining to call cancel operation where the calling UA sends a CANCEL request via openser to the called UA. After the UA sends ACK in response to 487 Transaction Terminated response, 487 responses are re-transmitted by openser to the UA as if the SIP proxy didn't regcognise the ACK request. I have tested with other UA and compare the differences between the ACK sent by UA that worked and UA which has this problem. The difference is that the ACK sent by the UA having this problem … [View More]

2 1

[Users] Accounting using dialog and sst module
by Asterisk Expert 03 Jan '07

03 Jan '07

Hello, I am currently trying to do accurate billing with openser. I found that dialog and sst modules can be used to determine failed transactions. i.e. If UA crashed and no BYE received at all. I tried to configure dialog and sst modules as specified in the documentation. But active dialogs keep incrementing even after hangup. It would be nice if someone can post a good example. I am using cvs head of openser. Snippet of config is as below: ###################### DIALOG ###################… [View More]

3 6

[Users] Help: how can I create one SIP account
by CESAR HUAMANI H. 03 Jan '07

03 Jan '07

Hello, Today I installed the openser in my debian etch since the CVS, I was looking information about how I can creat accounts SIP for my Endpoints (softphone and Hardphone ATA), please help me I'm new with the OPENSER Best regards

2 1

[Serusers] Detecting if username is in subscriber table
by Simon 02 Jan '07

02 Jan '07

Dear All, I would like to detect if a URI is for a user that is in the subscriber table. If I just use lookup("location") then I only detect those users that have registered. If a phone is switched off it is not registered and so will not be in the location table. However it will be in the subscriber table and I need to be able to check this from within the ser.cfg file to allow correct routing etc. Any Ideas ? Many Thanks Simon

2 1

[Users] Call forward record and Remove from callid in Openser
by Howard Tang 02 Jan '07

02 Jan '07

Hi All, I hope someone can help me with the following questions I have. 1. I have set up accounting module for record the call, simple outgoing call and incoming call are ok. However, if a sip phone activated the call forward, the call record doesn't show which sip client forward the call. e.g. A call B and B is the sip phone which forward the call to C. In the record, I can only see A call C. How can I see a record that B call C instead of A call C in the record? 2. I have connected to a … [View More]

2 1

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users