Hi,
How much of UAC package is implemented in LUA embedded module ? According to wiki seems like only "sr.uac.replace_from" is implemented. My goal is to "emulate" 3rd party registration by using UAC and LUA meaning that I will have to recreate register, tweak existing attributes and send it over to application server to process.
thanks
Dan
--
Daniel Ciprus
Integration engineer
http://www.acision.com
9954 Mayland Dr
Suite 3100
Richmond, VA 23233
USA
T: +1 804 762 5601
E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com>
________________________________
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while doing some tls tests, i noticed that if tls.cfg has a section like
this
[server:default]
verify_certificate = no
require_certificate = no
tls_method = SSLv23
private_key = /etc/sip-proxy/certs/sip-proxy/key.pem
certificate = /etc/sip-proxy/certs/sip-proxy/cert.pem
ca_list = /etc/ssl/certs/cacert.org.pem
then client does not give its certificate to kamailio server during tls
connection setup even if it had one.
if i specify:
require_certificate = no
then client sends its certificate to kamailio server, but if another
client does not have a client certificate, then it cannot connect at
all.
one way to solve this would be making kamailio listen on two tls ports,
one for clients that are required to present a a certificate and another
port for clients that do not have a certificate.
unfortunately, it is not possible to add a mask to ip address in tls.cfg
section like this:
[server:0.0.0.0/0:5062]
does anyone have a solution to this problem (other that running two
kamailio instances)?
-- juha
All,
I am facing problem with rewriting SDP part of the INVITE, in particular following is in my configuration:
#!subst "/RTP_INT_ADDR/10.96.173.15/"
#!subst "/RTP_EXT_ADDR/168.7.24.17/"
#!subst "/BOT_RTP_ADDR/10.96.173.52/"
subst_body('/RTP_INT_ADDR/RTP_EXT_ADDR/g');
subst_body('/BOT_RTP_ADDR/RTP_EXT_ADDR/g');
and this is what I ended up with:
c=IN IP4 10.96.173.15168.7.24.17
Apparently 2nd rule doesn't work rewrite, it's doing append instead. What I want to achieve is : if SDP part contains BOT_RTP_ADDR -> rewrite it. In case SDP part contains RTP_INT_ADDR-> rewrite it, so 2x if.
I have tried to use sdpops module, it's coring in my case and I have a reason to not upgrade, I just can't. Any other ideas how to handle my problem ?
Thanks
Dan
Server:: kamailio (4.2.0-dev0 (x86_64/linux))
Build:: mi_core.c compiled on 17:42:57 Dec 18 2013 with gcc 4.4.7
Flags:: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
GIT:: unknown
--
Daniel Ciprus
Integration engineer
http://www.acision.com
9954 Mayland Dr
Suite 3100
Richmond, VA 23233
USA
T: +1 804 762 5601
E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com>
________________________________
This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.
Given a proxy where the only registrations will be next-hop pbxes, where
those names will not be INVITEable, what is the ideal way to tell kama
the list of accepted usernames and authentication info for REGISTER?
It is important to ensure that any INVITE to any of the REGISTERable
usernames 404s.
Thanks,
-JimC
--
James Cloos <cloos(a)jhcloos.com> OpenPGP: 1024D/ED7DAEA6
Hello,
I'm wondering if we can set into < scan_prefix > an entrier with regex like
: 3[0-9]{3}?
If not, what would be the best module to manage in a DB the prefix that can
be composed by SIP clients before call cr_route? Something equivalent to a
dialplan I guess.
Thanks for your feedback.
Regards,
Igor.
Hi,
Not sure if this is a fair question. But how do you compare your call
processing engine against Cisco's Call Manager?
Appreciate your time.
Marjorie Lawrence
Global Wireless Networks, Inc.
Senior Voice Engineer/Unified Communications
Email: <mailto:Marjorie@gwnwireless.com> Marjorie(a)gwnwireless.com
Website: <http://www.gwnwireless.com/> http://www.gwnwireless.com
Office (301)674-5933
I have a scenario where calls that are transferred to a shared line doesn't trigger the "in use" light to show up on both phones that have the shared line.
but when i make an outgoing call, let's say, using the shared line, the "in use" light lights up on both phones that have the shared line.
Phone - polycom 650s.
Just a noob with SIP, kamailio and telecom, so not sure where to start debugging.
Any suggestions would be appreciated.
thanks in advance for your time.
Hello,
I try to set the "fr_inv_timer" dynamically from DB. Under 60s it works
fine. But I can't set more than 60s.
Here are the TM parameters:
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 5000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 60000)
Here are the instructions to set dynamically:
avp_db_load("$ruri/username", "$avp(s:inv_timeout)/usr_timeout");
t_set_fr($avp(s:inv_timeout)*1000);
Is there anything wrong?
Regards,
Igor.
I'm having some trouble with kamailio that uses the topology hider module and
an incoming call being put onhold by the caller. The INVITE containing the sdp
a=sendonly is to the 172.19.162.1 (the topoh mask_ip) contact header from the
200 OK to the initial INVITE.
U 109.235.36.178:5060 -> 109.235.32.40:5060
INVITE sip:172.19.162.1:5060;line=pcs-
mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*;user=phone
SIP/2.0.
This is handled within dialog (has_totag() && !loose_route()), which in the
default config ends in a sl_send_reply("404","Not here")
Explicitly handing INVITE (just relaying), results in the INVITE being sent to
172.19.162.1 (which doesn't exist)
U 109.235.32.40:5060 -> 172.19.162.1:5060
INVITE sip:172.19.162.1:5060;line=pcs-
mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*;user=phone
SIP/2.0.
How to get this INVITE to the actual destination? To get the same result for
the INVITE as for the ACK to the 200 OK?
U 109.235.32.40:5060 -> 109.235.36.178:5060
SIP/2.0 200 OK.
...
Contact: <sip:172.19.162.1;line=pcs-
mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy*>.
U 109.235.36.178:5060 -> 109.235.32.40:5060
ACK sip:172.19.162.1;line=pcs-
mp4KWiTsxRmtEAM3WGysx7y3xGnoxAxtuAxSuA7KEsktxGZKWgQfZD6smqlf9Gttzjy* SIP/2.0.
U 109.235.32.40:5060 -> 109.235.34.107:5060
ACK sip:+31756418030@109.235.34.107:5060;transport=udp SIP/2.0.
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Hello Everyone,
How I can specify ldap CA cert in ldap configuration ?
The directive ca_list bellow is ignored.
[07/Apr/2014:13:40:37 -0400] conn=20 op=-1 fd=70 closed - Peer does not recognize and trust the CA that issued your certificate.
[07/Apr/2014:13:40:37 -0400] conn=22 op=-1 fd=64 closed - Encountered end of file.
Current config
ldap_server_url = "ldaps://ds389.network.com"
ldap_bind_dn = "uid=adm,ou=People,dc=network,dc=com"
ldap_bind_password = "password"
authtype = simple
tls = on
ca_list = /etc/kamailio/cert/ca-cert_ldap.pem