Hello,
I'm trying to append a branch, assign it a q value so it has priority and
serialize the new branch and the main branch.
The goal is to send the new branch off to our CNAM provider, capture the
CNAM in their reply to a $var then drop the branch, stick the cnam in the
main branch and allow the call to continue as normal. We do this currently
using OpenSIPS and the serialize_branches(1) but the of load_contacts()
seems to be different than serialize_branches.
route[CNAM_DIPS]
t_on_branch("CNAM_DIPS");
t_on_reply("CNAM_DIPS");
t_on_failure("CNAM_DIPS");
t_load_contacts();
t_next_contacts();
append_branch("sip:2223334444@22.33.444.55:5600;trans-type=5", "0.5");
relay();
break;
Two problems. First when t_next_contacts is called I get the following
error:
ERROR: pv [pv_branch.c:58]: pv_get_branchx(): error accessing branch [0]
Second problem is that my call goes out on the branch I appended and I get
a reply from the CNAM provider but Kamailio forwards that reply back to the
phone causing the phone to send a second invite. In OpenSIPs we just called
drop, to drop the branch and let the main branch continue but this doesn't
seem to work in Kamailio.
I've read and reread the documentaion but it doesn't seem to behave the way
the documentation says it should. Is there something I'm missing? Is there
a better way to accomplish this in Kamailio?
Thank You,
___
John Petrini
Hello there,
I'm evaluating if I can use dial-plan and dispatcher module to build a
kamailio routing server to route calls based on number dialed(sometimes the
number dialed can be Alphanumeric).
I'm expecting have lot of dial-plan entries, some of them with
manipulations, others without.
Anyone can see here any limitation on this implementation?
One thing that I have noticed is that dial-plan module enforces me to fill
the fields substr_exp and repl_exp even if i don't have any manipulations
to apply.
I'm also concerning about performance issues with lot of dialplan entries.
What are your thoughts about this solution?
Thank you
--
José Seabra
Hi all
Follow scenario
Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> gateway kamailio [gw] --> carrier [carr]
I get Invites from [c5] with
Request ,To, from, contact, pid in national format 0794445566
[lbl] dispatches this to [gw]
For the [carr] I need international format.
So doing these transactions in [gw]
And sending to [carr] in international format
Request, to, from, contact, ... => 417794445566
Everything ok
Then I get a 100, 183 and even 200 from [carr]
Ack is coming from [c5] to [lbl] and [gw] - but then it stocks
The ACK is not sent to the [carr]
I kamailio log I see
DEBUG: RFC3261 transaction matching failed
DEBUG: t_lookup_request: no transaction found
So for me, the ACK cannot be assigned to a transaction and gets discarded by
if ( is_method("ACK") ) {
xlog(,"L_INFO", "WITHINDLG ACK - not loose route\n");
if ( t_check_trans() ) {
xlog(,"L_INFO", "WITHINDLG ACK - t_check_trans() \n");
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
t_relay();
exit;
} else {
xlog(,"L_INFO", "WITHINDLG ACK - not t_check_trans() DISCARD!!\n");
# ACK without matching transaction ... ignore and discard
route(NATMANAGE);
#t_relay();
#exit;
Any idea?
Problem with modifying the sip tags? Or problem with the dialog?
Thanks for helping
OIi
Hey all,
Back with more questions.
I'm using Kamailio to make an HTTP call to my API to perform authentication
and message routing. Currently, I'm trying to build up the post body that I
send to my API to make those decisions.
I've cherry picked a few of the headers that are important in my routing
decisions. But, ideally, I'd like to just iterate over all the SIP message
headers and append them as request parameters to my API call. Is this
possible with Kamailio? I've been looking through the docs and can't seem
to find a function to iterate the full list of headers.
Thanks in advance.
Best,
Colin
Good day! Can anyone point me to right direction? The question is - I
want to get list of registered users (to find out who is online at the
moment) from sip server. I'm using kamailio on server side and exosip
library on client side.
I tried to find something in RFC 3261 and failured... Can someone
help and tell what to do or where to look for the answer?
