Has anybody experienced having uac_replace_from() concatenate the old from:
value after the new domain?
for example:
From: "user1" <user1(a)domain.com>
becomes
From: "user2" <user2(a)domain.comuser1>
Kelvin Chua
Hi all,
i have a strange problem: i configured a basic IMS infrastructure
Kamailio-based, with a DNS server (bind9) running on a virtual machine with
IP address 192.168.100.55. Whenever i try to register a user, the first
registration almost always fails, while the second is successful. I
attached my dnszone config used with bind9 and the pcaps of the 2 regs; my
questions are
- is the IMS dnszone configuration correct?
- the problems i got are just due to latency issues or have i misconfigured
something?
Regards,
Andrea
HI All,
I am using the Kamailio 4.1.2 with RTPProxy 1.2.1 as a SIP Proxy. I want to
ensure that all my sip and media (RTP) passes through the SIP Proxy
(Kamailio + RTP Proxy) while we my SIP Client sends any SIP and RTP to SIP
Server (asterisk). So it is like below:
SIP Agent <--------SIP+RTP---------->SIP
Proxy(Kamailio+RTPProxy)<------SIP+RTP-------->SIP Server(Asterisk).
Now it is working fine with my current configuration on Kamailio with
RTPProxy with SIP and RTP both if my SIP Agent is on the 802.11 LAN that is
my PC have local IP address like 192.168.1.2.
Now I am facing the problem in the case where my PC is accessing the
internet through the USB data card (or Internet USB dongle). Here my PC IP
is like 116.203.51.209. In this case my SIP is successfully passes through
the SIP Proxy and works good. But in this case my RTP is does not passes
through the SIP Proxy (RTPProxy). The SIP Proxy(Kamailio + RTPPRoxy) and
SIP Server is running on the public IP.
Please find attached my kamailio.cfg.
Please let me know what is the issue.
Thanks and Regards
Varun
Hi,
I am at the point where connection is established and no apparent errors are reported.
However audio is not output.
The rtp traffic seems to be transfering between the points as conclueded because Asterisk debug log shows
Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021868, ts 221760, len 4294967284)
Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001383, ts 1917269534, len 000160)
Sent RTP packet to 10.1.xxx.xxx41143 (via ICE) (type 08, seq 021869, ts 221920, len 4294967284)
Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001384, ts 1917269694, len 000160)
Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021870, ts 222080, len 4294967284)
Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001385, ts 1917269854, len 000160)
And the browser machine on the other endpoint on a tcpdump does shows traffic on the port (41143)
What could be causing there to be no audio?
This is the connected sdp
=0
o=root 350315728 350315728 IN IP4 10.31.xxx.xxx
s=Asterisk PBX 12.2.0-rc1
c=IN IP4 10.31.xxx.xxx
t=0 0
m=audio 24316 UDP/TLS/RTP/SAVPF 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:1c5c5d52130f06fd70e1e23f0d6323f2
a=ice-pwd:12611b8146599a9019d59b4b649a7970
a=candidate:Ha1f026f 1 UDP 2130706431 10.31.xxx.xxx 24316 typ host
a=candidate:Ha1f026f 2 UDP 2130706430 10.31.xxx.xxx 24317 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8
a=sendrecv
Hi all,
i am trying to configure an IMS Core Kamailio-based: while configuring the
P-CSCF, i set up the parameters 'db_url' and 'db_mode' of the module
ims_usrloc_pcscf respectively to
'mysql://kamailio:kamailiorw@localhost/kamailio'
and 1 (because i want a DB for persistence). I had created successfully the
DB using kamdbctl, and loaded the module 'db_mysql' in kamailio.cfg. Then i
tried to run Kamailio, and i got this error:
ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query: driver error on
query: Unknown column 'aor' in 'field list'
What am i missing? Consider then when i created the DB i answer yes to all
the questions about creating additional tables ...
Andrea
Hi,
When using sql_xquery() like this:
sql_xquery("ca", "SELECT * FROM gateways", "gateways");
... what's a good way to check if any rows were returned? Since one does
not have a $dbr(gateways=>rows) value in this scenario, what should one do?
- is_avp_set("$xavp(gateways=>id)")) does not appear to operate on
XAVPs, or at least, the fixup functions reject them:
ERROR: avpops [avpops.c:935]: fixup_is_avp_set(): bad attribute name
<$xavp(gateways=>id)>
- the 'defined' operator does not appear to return a negative condition
here:
if(!defined $xavp(gateways=>id))
This condition evaluates to true.
Much appreciated!
-- Alex
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Hi all
We use kamailio 3.3.7 and rtpproxy for enduser call-termination.
In case of a fax call, we get an invite from our carrier for codecs G711a/u and T38.
As our termination carrier does not support T38 and because the invite contains G711 and T38 we get back error 488.
How is it possible to remove the whole T38 part of this invite?
We tried
sdp_remove_codecs_by_name(list) without success - what "name" should we use for T38? [T38, t38, t.38, T.38, ...]
sdp_remove_line_by_prefix(string)
sdp_remove_media(type)
None of these functions did really work - best was the last one with type=image but then the sip header is malformed.
As we saw with Kamailio version 4.1.x there are a lot of new functions within sdpops. Would an upgrade help?
So basically the question is:
How to remove the t38 part of the fax invite? (see attachment)
KR,
Oli
[cid:image001.png@01CF523C.988433F0]
Hello everyone,
>From today i started getting SIP 480 error when making outbound call.
I checked out with a different voip provider and got SIP 500 error.
Internet calling is working fine.
The service (outbound calls) was working fine till yesterday.
I'm a newbie to kamailio and voip, so need your help.
Please guide me to resolve this error.
Thanks in advance.
Hello,
I don't know a lot about SIP with TCP, so I thought I'd ask for some
opinions here. I've been testing Kamailio with TCP against the Android
SIP client (4.4.2) and have found, to my annoyance, that it sends a RURI
with a transport attribute in its INVITEs:
INVITE sip:14045551212@sip.evaristesys.com;transport=tcp
My upstream gateways are all UDP, so this causes Kamailio to attempt to
contact them all via TCP (and fail).
I know how to bend the transport with Kamailio, and I can strip this
attribute. My question is more methodological: is this "correct" for a
client to do?
In my opinion, the answer is no. The client should not be so
presumptuous. It should specify the transport in its Contact, to
indicate how it wants to be reached, and leave other decisions about
transport to the UAS. But is there something I am perhaps missing?
Thanks,
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Greetings.
I have the next problem:
iOS based clients connect via TLS to kamailio server.
They run mostly in background mode - it means connection refresh interval
is ~10 minutes.
Some of clients reside behind paranoidal routers which considers such idle
connections as lost and closes them.
I see only way to resolve it is to send OPTIONS/NOTIFY/CRLF from kamailio
to clients.
Tried to use nat_traversal module by calling nat_keepalive() to every
REGISTER message. But there is no incoming keepalive (OPTIONS or NOTIFY) on
client side.
Is there another way to make "heartbeat" in SIP/TLS from server side?
Thank you.
