Hi Peter,
Still I am unable to make call to Jitsi to Jitsi. Call established But no
media. But text message works.
NAT traversal work with RTPProxy and I can do IPPhone to IPPhone call and
voice works very well.
Tried with many soft phones and voice did not go through.
Did I miss something in the configuration ?
Please advice me.
Best Regards,
Roy.
Andrew,
Congratulations on your new position as a Kamailio developer! When I saw
your module (SCA) I began testing immediately since it may be a stepping
stone to help me get the Polycom barge-in feature working with Kamailio.
However, I ran into a few difficulties. Can you assist me?
To make your existing code work as a K module I had to change a few lines
of code and recompile under modules_k. Specifically, since the K definition
of the ucontact structure defines ucontact->aor as str* and the S
definition as str**, I changed “&c->aor” to “c->aor” in the following lines
of sca_usrloc_cb.c:
73: if ( !sca_uri_is_shared_appearance( sca, c->aor )) {
74: LM_DBG( "%.*s is not a shared appearance line", STR_FMT( c->aor ));
79: &SCA_EVENT_NAME_CALL_INFO, c->aor ) < 0 ) {
Since I use PostgreSQL as my SQL engine I created the SCA entries using
this code:
INSERT INTO version (table_name, table_version) values
('sca_subscriptions', '0');
CREATE TABLE sca_subscriptions (
id SERIAL PRIMARY KEY NOT NULL,
subscriber VARCHAR(255) NOT NULL,
aor VARCHAR(255) NOT NULL,
event INTEGER NOT NULL,
expires INTEGER NOT NULL,
state INTEGER NOT NULL,
app_idx INTEGER NOT NULL,
call_id VARCHAR(255) NOT NULL,
from_tag VARCHAR(128) NOT NULL,
to_tag VARCHAR(128) NOT NULL,
notify_cseq INTEGER NOT NULL,
subscribe_cseq INTEGER NOT NULL,
CONSTRAINT sca_subscriptions_idx UNIQUE (subscriber, call_id, from_tag,
to_tag));
I applied the recommendations from your documentation on configuring
Kamailio (3.3.1) and inserted the call to “sca_handle_subscribe ()” in
kamailio.cfg.
Next, I connected two Polycom phones (SP335 and SP650) to my network and
configured them to enable presence and use a shared line. Four lines were
configured on the phones as shown below:
SP335 Line 1 (p11) = private line 6013
SP335 Line 1 (p12) = shared line 1519
SP650 Line 1 (p21) = private line 6014
SP650 Line 2 (p22) = shared line 1519
My tests get inconsistent results on the line LEDs and it seems that they
do not work as a shared line. When I ran a tcpdump of the SIP communication
(see attached pcap files) I found several anomalies:
- Some packets (sca-reboot1 #22, 44, 113, 120 and sca-reboot2 #17, 38)
contain spurious NUL characters in the tags section. In all tests it seems
to occur in the Expire tag when Kamailio accepts the subscription.
- When Kamailio sends call-info NOTIFY the Content-Length tag is missing.
- Sometimes the phone does not respond to a call-info NOTIFY event (
sca-reboot1 #23, 45 and sca-reboot2 #18, 39).
- I don't have an example here but sometimes the phones respond to the
call-info NOTIFY with "481 Call Leg/Transaction Does Not Exist".
I think some of my problems are related to these anomalies. Could you help
me to determine if this is a problem with the code itself, my failure to
properly port it to the K module, or some other issue?
Once again, thank you for making this very useful module for Kamailio and
for the documentation to make it work. I hope that resolving my problems
may help the code to become a part of Kamailio.
Sincerely yours,
Bob Boisvert
I am trying to get Linphone to show presence information. I did my Kamailio setup using just the template from a stardard install and it is working properly with Bria (X-Ten). However, when using Linphone it does not show buddies presence. Any hint? Cheers! Moacir
All the topics/examples I found on the net is not what I am looking
for, so here is the question: How do I achieve parallel forking (call all
devices) if I have a single user registered using the same account from different
devices (i.e.: table phone, iPad and PC)? What I want is to ring all devices a
user may be using. Cheers! Moacir
core cookbook reads:
append_branch
Similarly to t_fork_to, it extends destination set by a new entry. The
difference is that current URI is taken as new entry.
what is t_fork_to? i could not find it in tm README.
-- juha
Hi,
I am running Kamailio 3.4 with web-socket and everything works fine with
NAT traversal for IP Phones.
But when I enable the PRESENCE it crash with "(3) errors in config file"
I have copied the config.cfg to paste-bin at
http://pastebin.ca/2257817<http://pastebin.ca/2257815>
Please guide me on this issue.
Best Regards,
Roy.
Dear All
I ma trying to have all SIP signalling through TLS. Using Kamailio 3.1.5.
In a typical setup, endpoints initiates call, messages reach Kamailio
proxy, Kamailio proxy forwards these packets to main proxy.
In Kamailio.cfg, I have added below code.
route {
# per request initial checks
route(REQINIT);
setflag(FLT_NATS);
if(is_method("REGISTER"))
{
t_relay_to("tls:115.114.48.19:443");
exit();
}
else
{
xlog("incoming request\n");
route(WITHINDLG);
route(RTPPROXY);
t_on_reply("REPLY_ONE");
t_on_failure("FAIL_ONE");
t_relay_to("tls:115.114.48.19:443");
exit();
}
..............
I am finding most of the messages are getting transmitted in TLS. But few
ACK and REFER / REFER response packets are still being transported in UDP.
Can somebody tell me whats the wrong I am doing here.
By any chance
route(WITHINDLG);
route(RTPPROXY);
before t_relay_to("tls:115.114.48.19:443");
can cause this ????
And one more doubt.., 115.114.48.19 I get in route header. How can I
extract the value of route header, so that I do not have to hard-code
Many thanks in advance.
-kamal
Hello
I'm trying to execute a Lua script on kamailio startup. I'm calling the
script inside event_route[hable:mod-init], but it fails to be executed
with this error:
[app_lua_mod.c:167]: Lua env not initialized
It happens both with lua_run and lua_dofile.
I've played with the order in wich the modules are loaded, but I haven't
been able to get rid of that message. Kamailio version is 3.3.0. Do you
think this is feasible or should I try a different solution?
Thanks in advance
Regards,
Vicente.
Hi All,
I've been trying to get some Cisco/Linksys SPA phones working with Kamailio (current stable). All seems to be ok apart from getting Presence/BLF/SCA working.
The phones are set to
Server_Type : RFC3265_4235
Line Key 2 Extension : Disabled
Line Key 2 Share Call Appearance : Private
Line Key 2 Extended Function : fnc=blf+sd+cp;sub=1001@ourserver.com;ext=1001@ourserver.com
Where 1001 is the other phone.
The phone appears to be sending event type 'call-info' rather than dialog etc.
Phone is being sent '489 Bad Event' and the error in the logs is : presence [subscribe.c:1007]: Unsupported event header field value call-info
Whatever I set the Server Type to on the phones it still tries to send call-info events on subscribes
Has anyone got Presence working with Cisco SPA handset?
The module needed seems to be available on OpenSIPs
http://www.opensips.org/html/docs/modules/1.8.x/presence_callinfo.html
Thanks!
Mark
Anybody got a good tutorial or starter project for Kamailio?
We have Siremis 3 installed, but there is 0 help on that?
Trying to solve 2 scenarios:
-asterisk to softphone through kamailio
-softphone to asterisk through kamailio.
I ultimately want to use Kamailio as an edge router
Any help is appreciated.
Matt Scott
