Hello,
First, i have 7 years experience with Asterisk, but I started a project
with Kamailio, forgive me in advance if I say silly things...! ;-)
I set up a classic Asterisk / Kamailio configuration:
sip phones -> kamailio -> asterisk -> sip trunks/pstn.
When a call comes from the PSTN side, if I configure Asterisk as follows:
[012345678]
type = friend
username = 012345678
secret = xxxxxxx
host = dynamic
fromdomain = sip.mydomain.com
fromuser = 012345678
Standard mode:
exten => …
[View More]012345678, 1, Dial(SIP/012345678) -> The call is redirected on the
phone by Kamailio ! :-)
------------------------------------------------------------------------------------------------------------------------------------------------
Trunk mode:
[mytrunk]
type = friend
username = mytrunkUser
secret = xxxxxxx
host = dynamic
fromdomain = sip.mydomain.com
fromuser = mytrunkUser
exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected
by Kamailio....
exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected
by Kamailio ....
My question is how to allow the routing of multiple numbers (trunk mode) in
a SIP account with Kamailio?
Best regards,
Mickael
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Hello,
We are running Kamailio 4.2.6 ( db77ac) compiled with memory debugging.
The free PKG memory has decreased on ~6.5 M after two weeks.
Please see attached file with the log messages at kamailio shutdown.
Thank you in advance,
Julia.
Hi list,
Record-route on sequential requests doesn't have did/vsf parm.
On initial INVITE, I call record_route() and set dialog flag.
kamailio adds the following RR
Record-Route: <sip:ww.xx.yy.zz;lr;ftag=15af612df;vsf=AAAA....;did=4a8.3ca2>
UAC sends a re-INVITE, containing the following route:
Route: <sip:178.21.251.54;lr;ftag=15af612df;vsf=AAAA....;did=4a8.3ca2>
I call loose_route() followed by record-route(), and kamailio adds the following RR:
Record-Route: <sip:ww.xx.yy.…
[View More]zz;lr;ftag=15af612df>
I've got a phone which updates the routeset on sequential requests (which I don't think it should) and the did-matching fails.
Should kamailio add did/vsf on sequential requests, when I call record_route()?
Should I set dlg_match_mode to 1? (I use 0)
Is it wrong to call record_route() from sequential requests?
Which way is the best to resolve the problem?
Regards,
Kristian Høgh
Uni-tel A/S
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Hello everyone,
this is my first message. I'm glad to become a part of this big community.
I'm new in SER technology. I worked in Asterisk PBX only, until now.
For the first time i installed Elastix MT witch have kamailio in front of
asterisk, and now i have a problem.
I attach a diagram of initial configuration in my lab and pcap file and
kamailio.cfg.
Regardless diagram, x.x.x.x addresses are public and y.y.y.y are private.
When i test connection 1.1 is working,
connection 1.2 not working
…
[View More]connection 2.2 not working
connection 1.3 not working
connection 2.3 working
Phones are connected(authenticate) to kamailio, but when i look at asterisk
cli those phones are unreachable, but can call, but can not be called.
On router where android works, Siemens dont, and where pc softphone works
other UAC dont work.
I think that my nat-traversal configure good, regardless of documentation.
Please tell me, if someone has some ideas of solve this problem.
Thanks in advance, and sorry for my bad english :)
----------------------------------------------------------------------
*Vladimir Stepić,*
Engineer specialist in computer networking
System Linux Administrator
mob. tel. +381 69/18-68-431
email: e1.e398(a)gmail.com
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Hi,
I'm doing some experimentation on kamailio's udp_server.c. Now i need
libpcre for that. So i first thought if i enable the regex module, the
CFLAGS/LDFLAGS would be available from udp_server.c. But it turns out
it doesn't build(doesn't *link* to be more precise).
Kamailio's build system is pretty unfamiliar for me. Although i
absolutely hate autotools based build system(I prefer Scons), but i'm
pretty familiar with these. But Kamailio's build system is pretty much
cryptic for me(Is it …
[View More]based on CMake?).
There is a page on devel wiki for kamailio's build system. it turns
out i can do stuff like :
make cfg prefix=/opt/kamailio \
SCTP=1 \
CC_EXTRA_OPTS=-I/usr/gnu/include \
group_include="standard postgres presence" \
include_modules="snmpstats lcr tls"
So there might be a LD_EXTRA_OPTS or stuff like that. Which than would
be defined (I guess) globally so the core c files(like udp_server.c)
would be able to use that. One way i guess would be to sharpen my
egrep and find that variable name. But it all feels a bit hacky. As i
have already included regex module, there should be a way to reuse the
CFLAGS/LDFLAGS for that module for core files like udp_server.c.
Cheers
Arif
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Hi All.
Does anyone know where to find training guide for Kamailio HA? I want to have 2 Kamailio in HA with four (4) Freeswitch servers behind them. I will also like to customer cgrate to manage the accounting side in a multi-tenant environment. I am just researching for this on my own and I will appreciate a guide to realize this objective.
Thanks
Hello,
I have rebooted my kamailio machine and now the value of kamailio dialogs
is always 0:
KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0
KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0
KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0
KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0
And also the status is:
KAMAILIO-MIB::kamailioDialogUsageState.0 = INTEGER: idle(0)
How can I change it to active(1)? I have dialog.so loaded as well as
modparam("…
[View More]dialog",
"dlg_flag", 4) and dlg_manage().
Thank you.
Regards,
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Daniel,
I got the following error message (currently running version 4.3):
Nov 27 18:56:51 ip-10-1-2-100 /usr/local/sbin/kamailio[451]: ERROR:
dispatcher [dispatcher.c:517]: mod_init(): call load dispatching DST
ID_AVP set but no size for hash table
Nov 27 18:56:51 ip-10-1-2-100 /usr/local/sbin/kamailio[451]: ERROR: <core>
[sr_module.c:962]: init_mod(): Error while initializing modul
e dispatcher (/usr/local/lib64/kamailio/modules/dispatcher.so)
I noticed in the following DEV GIT (…
[View More]Version 3.1) message that the error
message was changed to a message that more clearly defined the error and
correction for it. But the my current message would indicate that it got
changed back to the older error message.
http://lists.sip-router.org/pipermail/sr-dev/2010-November/009799.html
Just thought I would point it out because the newer message from the GIT
message seems clearer to me.
--
*Sam D Ware*
CIO
*Vera Networks LLC*
Email: sam.ware(a)veranetworks.com
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