Re: [OpenSIPS-Users] Does OpenXCAP support IPv6 network?
On 11/15/2010 04:23 AM, CheeWii wrote: Hi, Does OpenXCAP support IPv6 network? How to let OpenXCAP listen an IPv6 address? OpenXCAP hasn't been tested with IPv6 as of today. Adjustments will be necessary most probably. -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Failover route with drouting on OpenSIPS 1.5.X
Hello to all members. I have found a tutorial for configuring failover routes on OpenSIPS, and I see it is defined for module in OpenSIPS 1.6.X; Is it possible to have such feature for OpenSIPS 1.5.X? Thanks in advance for your attention. Best regards. -- Sergio Gutiérrez ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] No ACK response for 200 ok
Hello!!
I m new in opensips and i m testing the load balancer cause i need it to
balance calls between 4 asterisk.For the start i make the following
scenario
Cisco gateway inbound -- opensips -- asterisk -
Cisco gateway outbound
when the call comes to the opensips, the load_balancer forward the call
correctly to my asterisk but the call hangs up after 15 seg
approximately.When i did a ngrep for the sip traffic in opensips, i
realized that cisco gateway inbound never sent the ACK for 200 OK to
opensips .
In the Cisco's logs i saw that the reply of 200 ok is sent directly to
public ip of asterisk but never to opensips server so asterisk still waiting
for the ACK from opensips.
In the same way opensips never receive the BYE packet and the load never
decrease when the call is hanging up.
Cisco gateway opensipsasterisk
---invite---
--trying ---invite---
---trying---
200OK---
---200 OK---
200OK---
---200 OK---
200OK---
---200 OK---
200OK---
---200 OK---
Please can somebady help me to understand what cause that?
Best Regards!!
Configuration of Cisco AS
...
dial-peer voice 205 voip
description VoIP GWs-OPENSIPS LAB Desarrollo
preference 1
destination-pattern ^91112666[7-9]$
session protocol sipv2
session target dns:x$s$.$d$.opensips.lab.egtelecom.es
dtmf-relay sip-notify rtp-nte
no vad
.
This my opensips.cfg:
debug=4
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=4
disable_dns_blacklist=yes
auto_aliases=no
port=5060
### Modules Section
mpath=//lib/opensips/modules/
loadmodule db_mysql.so
loadmodule signaling.so
loadmodule auth.so
loadmodule auth_db.so
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
loadmodule registrar.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule uri.so
loadmodule acc.so
loadmodule dialog.so
loadmodule load_balancer.so
loadmodule presence.so
loadmodule presence_xml.so
# - setting module-specific parameters ---
# - mi_fifo params -
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
# - rr params -
# add value to ;lr param to cope with most of the UAs
modparam(rr, enable_full_lr, 1)
# do not append from tag to the RR (no need for this script)
modparam(rr, append_fromtag, 1)
# - registrar params -
/* uncomment the next line not to allow more than 10 contacts per AOR */
modparam(registrar, max_contacts, 10)
# - usrloc params -
modparam(usrloc, db_mode, 0)
/* uncomment the following lines if you want to enable DB persistency
for location entries */
modparam(usrloc, db_mode, 2)
modparam(usrloc, db_url,mysql://opensips:opensip...@localhost
/opensips)
# - uri params -
modparam(uri, use_uri_table, 0)
# - acc params -
/* what sepcial events should be accounted ? */
modparam(acc, early_media, 1)
modparam(acc, report_ack, 1)
modparam(acc, report_cancels, 1)
/* by default ww do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable append_fromtag
in rr module */
modparam(acc, detect_direction, 0)
/* account triggers (flags) */
modparam(acc, failed_transaction_flag, 3)
modparam(acc, log_flag, 1)
modparam(acc, log_missed_flag, 2)
/* uncomment the following lines to enable DB accounting also */
modparam(acc, db_flag, 1)
modparam(acc, db_missed_flag, 2)
# - auth_db params -
/* uncomment the following lines if you want to enable the DB based
authentication */
modparam(auth_db, calculate_ha1, yes)
modparam(auth_db, password_column, password)
modparam(auth_db, db_url,mysql://opensips:opensip...@localhost
/opensips)
#modparam(auth_db, load_credentials, )
# - dialog params -
modparam(dialog, dlg_flag, 13)
modparam(dialog, db_mode, 1)
modparam(dialog, db_url, mysql://opensips:opensip...@localhost
/opensips)
# - presence params -
/* uncomment the following lines if you want to enable presence */
modparam(presence|presence_xml,
db_url,mysql://opensips:opensip...@localhost/opensips)
modparam(presence_xml, force_active, 1)
modparam(presence, server_address, sip:10.234.227.199:5060)
# - load_balancer params -
modparam(load_balancer, db_url,mysql://opensips:opensip...@localhost
/opensips)
### Routing Logic
# main request routing logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,looping);
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
[OpenSIPS-Users] nat transversal
hey, actually i wanted to know how the nat helper module manage to inform the rtpproxy of the port number used by a symmetric nat, cause in the sdp packet, the only thing that is mentionned is the public address of the nat, and not the port number used by the nat. thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Next OpenSIPS release - planing
Hi all, Regarding the next opensips release, the near future plan is : 1) Stable release - 1.6.4 Estimated: mid December TODO : start testing all new stuff from trunk and port them to 1.6 branch solve bugs and patches from SF tracker 2) Code upload on git - opensips 2.0 Estimated: before Christmas What will be able to do: stateless SIP routing based on DNS/IP (routing part is not yet developed). What will be in there: async reactor async transport layer (UDP and TCP) async parser (with partial parsing and resume parsing capabilities) async DNS resolver async DB support (mysql and postgres) message changing (new version without lumps) Let me know if there are any topics to be discussed related to these two code releases. Best regards, Bogdan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Failover route with drouting on OpenSIPS 1.5.X
Hi Sergio, Sure you can do it with 1.5 also. Regards, Bogdan Sergio Gutierrez wrote: Hello to all members. I have found a tutorial for configuring failover routes on OpenSIPS, and I see it is defined for module in OpenSIPS 1.6.X; Is it possible to have such feature for OpenSIPS 1.5.X? Thanks in advance for your attention. Best regards. -- Sergio Gutiérrez ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Timer Based Failover Question
I do this.. Works really well..
-Brett
On Tue, Nov 16, 2010 at 12:49 PM, Dave Singer dave.sin...@wideideas.comwrote:
I ran into the same problem with one of our carriers. The way I did it,
with advice from bogdan, was to set the fr_inv_timer_avp to 6 sec (so long
because of some calls to cell phone systems have long delays) when sending
to the particular carrier then in a reply route special to that carrier,
reset it to the normal 300 if the response was a 18x. Here are
the pertinent parts:
modparam(tm, fr_inv_timer, 300) # Timer on Final response:
Minimum is 2 sec, Default is 120 sec.
modparam(tm, restart_fr_on_each_reply, 1) # Reset fr_int_timer on
each reply. Needed if you want to adjust the fr_inv_timer_avp with avp
depending on reply.
modparam(tm, fr_inv_timer_avp, $avp(i:2)) # Used if overide of
fr_inv_timer param is needed.
modparam(tm, onreply_avp_mode, 1) # set to 1 if you want to
access and or save avps from or for other parts of the transaction, like
changing the fr_inv_timer_avp.
route[carrier_c] {
if (is_method(INVITE)) {
t_on_failure(2);
t_on_reply(2);
$avp(i:2) = 6;
}
}
onreply_route[2] {
/* once we get ring progress let it ring for upto 300 sec */
fix_nated_contact();
if ( $rs =~ 18. ) {
$avp(i:2) = 300;
#xlog(got ringing, reset final timer to $avp(i:2)
sec.\n);
}
}
On Tue, Nov 16, 2010 at 2:49 AM, Denis Putyato denis7...@mail.ru wrote:
And what about
http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id250384
*From:* users-boun...@lists.opensips.org [mailto:
users-boun...@lists.opensips.org] *On Behalf Of *Bruce Borrett
*Sent:* Tuesday, November 16, 2010 1:40 PM
*To:* Users@lists.opensips.org
*Subject:* [OpenSIPS-Users] Timer Based Failover Question
Hi All
I am having a problem where a SIP provider are sometimes sending us a 100,
but then nothing afterwards. I would like to fail these calls over using a
timer, but fr_timer wont work since we are receiving a 100, and fr_inv_timer
requires a very lengthy duration which also will not work as I would like
for the call to failover within 5 seconds maximum.
Does anyone have any other suggestion for me please?
Regards,
Bruce Borrett
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