sr-users November 2009

sr-users@lists.kamailio.org

80 participants
108 discussions

[Kamailio-Users] openser freeradius connectivity error

by toqeer ali

Hi all, I configured freeradius for accounting of openser. but when i make a call the call establishwa and gives this error for Freeradius. Nov 18 06:50:58 kamailio /sbin/openser[11590]: rc_ip_hostname: couldn't look up host by addr: 0A000005 Nov 18 06:50:58 kamailio /sbin/openser[11590]: rc_send_server: no reply from RADIUS server unknown:1813 Nov 18 06:50:58 kamailio /sbin/openser[11590]: ERROR:acc:acc_rad_request: radius-ing failed Nov 18 06:51:20 kamailio /sbin/openser[11593]: rc_ip_hostname: couldn't look up host by addr: 0A000005 Nov 18 06:51:20 kamailio /sbin/openser[11593]: rc_send_server: no reply from RADIUS server unknown:1813 Nov 18 06:51:20 kamailio /sbin/openser[11593]: ERROR:acc:acc_rad_request: radius-ing failed please help -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

14 years, 5 months

[Kamailio-Users] Play Early Media

by alex pappas

Dear all, I'm trying to play Early Media instead of Ringing WHEN an 180 Ringing OR 182 Call Queue message is received from the destination UAC. In the onreply_route[1] I added the following lines in bold: I use kamailio 1.5 and I have NAT enabled. onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"*(180)|*(183)|(2[0-9][0-9])") { force_rtp_proxy(); } * if (status=~"(180)") { rtpproxy_stream2uac("/usr/local/etc/rtpproxy/ast-ivr.alaw",""); } else { rtpproxy_stop_stream2uac(); }* if (isbflagset(6)) { fix_nated_contact(); } } When I get from a UAC an 180 Ringing does not play the media and also in the syslog I see the following ERRORs: Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:extract_body: message body has length zero Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:force_rtp_proxy: can't extract body from the message Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:select_rtpp_node: script error -no valid set selected Nov 17 14:11:29 YGOR /usr/local/sbin/kamailio[4681]: ERROR:nathelper:rtpproxy_stream: no available proxies Thanks in advance for any help Alex

14 years, 5 months

[Kamailio-Users] Dialplan Module Confusion

by JR Richardson

Hi All, I'm new to the list and fairly new to the vastness which is kamailio and the tons of modules and options. I'm trying to figure out the simplest method to accomplish the following scenario: SIP call comes in, match DID in database (MySQL), rewrite ruri with associated database entry matching DID, relay call to specified gateway, if no matching entry for DID is in database, forward call to dispatcher module and continue processing call (I already use the dispatcher module for load balancing, just need to implement DID call routing before I send calls to dispatcher) I've been reading about the following modules, dialplan, LCR, drouting, alias_db. I think dialplan is what I really need, but I'm confused about what entries to put in the database to match on 10 digit DID: match-op = 0 (equal) match_exp= 1112223333 match_len= 10 subst_exp= ?? is this where I would put the ruri sip:1112223333@10.10.10.10<sip%3A1112223333(a)10.10.10.10>(I'm not sure of the syntax here) repl_exp= ?? not sure what this is for can more than one database entry have the same id? and also I could use a practical example of how this would work int he config file: dp_translate("240", "$ruri.user/$ruri.user"); t_relay?? Any guidance on this would be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

14 years, 5 months

[Kamailio-Users] RADIUS for billing

by Rajesh

Dear All, I am trying to configure openSER for billing.After a lot og googling, i came to know that RADIUS is needed for anabling billing module. I would like to know if RADIUS is a MUST for billing or can i manage using mysql alone. I have configured accounting module in openser.can i use mysql and a CDR tool for billing?.. -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

14 years, 5 months

[Kamailio-Users] dbtext, domain and caching

by Andreas Granig

Hi, I was trying to use caching in domain module with the dbtext backend, but on startup I get the error ERROR:domain:reload_domain_table: Database problem Looks like something is wrong with the column type when it's checked in line if ((ROW_N(row) == 1) && (VAL_TYPE(val) == DB_STRING)) { ... } The db file for domain module looks like: id(int) domain(str) last_modified(int) 1:test.local:0 Any hints on that? Probably a problem with dbtext module, since it works with mysql? Best regards, Andreas

14 years, 5 months

[Kamailio-Users] Test a string concatenated with an AVP

by Antonio Reale

Hi all, actually I test if the R-URI contains the value *44* with: if (uri=~"^sip:\*44\*[0-9]+@") { ... } Now I want to read the value 44 or *44* from an AVP. Tried using directly the avp or assigning it to a variable, without success. ex.: $var(x) = "sip:" + $avp(s:code) + "[0-9]+@"; How can I solve this problem? Thank you very much. Regards. -- Antonio

14 years, 5 months

[Serusers] (no subject)

by Chahn Kim

Hi all, Need some help from experts. I have beeb using $hdr(via[0]) to fetch the VIA header to get top via field, which will show entire top via filed as; <SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK-9aa0fded>. Now I am in need of getting IP:port# portion only to get assigend to a variable. What would be the most effecient way of doing it? I didn't find any other pseudo variable to provide it based on docu. Thanks in advance. John Chahn Kim email: johnk(a)koreanet.us

14 years, 5 months

[Kamailio-Users] Question about force_send_socket() as it relates to record_route()

by Geoffrey Mina

Hello, I have a scenario where I need to send traffic to my carrier from two different source IP addresses. The carrier charges me one rate for dialer based traffic and another rate for standard outbound origination. I am planning on using the force_send_socket() method to manually set the source IP address. My question is how does this correlate to my usage of the record_route() function? Do I need to force_send_socket() BEFORE inserting the Record-Route header? Are there any other things I need to be concerned with while using the TM module? Thanks!

14 years, 5 months

[Kamailio-Users] Freeradius Configuration

by toqeer ali

I am configuring Freeradius server with openser... By default there is openser dictionary file that the path of that file should be included freeradius master dictionary file... now i am confuse that command $INCLUDE {path of the openser dictioanry file} how i will include that over the network in dictionary file of Freeradius... if i copy the Freeradius server then it gives error... Basically my radius and openser are both on diffrent servers. -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

14 years, 5 months

[Kamailio-Users] Possible setups question

by Varga Levente

Hello! I'm new to Kamailio so I want to ask a few questions before to be sure that I'm not trying to do something that cannot be done using Kamailio. The question would be if it is possible to do the following things using Kamailio: I want to setup the following configuration, two or more Asterisk servers with one redundant server in standby using Kamailio to route and partition the number. of users in a fixed manner. What I mean by that for instance if I have 200 users I want that the first 100 user to register and use server number one and the other 100 user to use the server number two. A user cannot migrate from one server to another they have to be always on the designated server from start. Kamailio would act as a unified front-end/gateway for the client (they will use the same entry point not being aware that there a number of Asterisk server beside it). Also there is a need for a N+1 redudancy scheme, that is if one of the servers fail or get very busy the new calls and registration should be redirected to the standby server on the fly (user account data replication would be taken care of on the spot). Can all this be done using Kamailio as a gateway and SIP router? Thank you! levi -- Lege, lege, relege, ora, labora et invenies.

14 years, 5 months

← Newer
1
...
4
5
6
7
8
9
10
11
Older →

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 November 2009