Hello
Getting this issu on Kamaiio 5.0 ( install from GIT):
Apr 10 10:47:42 ns3033789 /usr/local/sbin/kamailio[11496]: WARNING: rtpproxy [rtpproxy.c:1366]: rtpp_test(): can't get version of the RTP proxy
Apr 10 10:47:42 ns3033789 /usr/local/sbin/kamailio[11488]: WARNING: rtpproxy [rtpproxy.c:1403]: rtpp_test(): support for RTP proxy <udp:127.0.0.1:7722> has been disabled temporarily
I done install of RTPproxy with:
$ git clone -b master https://github.com/sippy/rtpproxy.git
$ git -C rtpproxy submodule update --init --recursive
$ cd rtpproxy
$ ./configure
$ make
$ make install
$ rtpproxy -l 5.135.177.203 -s udp:localhost:7722
Before i install mysql-server & kamdbctl create (rtpproxy table), it was working fine
Someone to advise please!
Keffa Y. Diomande
Skype : keffadi
Tel : +225 02 77 22 22
Office : Abidjan Ciad
http://www.alcall.net
Hello,
just to let everyone know that I invited Mikko Lehto into the developer
group on Kamailio gihub project. He has made a lot of good pull requests
over the time, specially to documentation and unit tests that could have
been just committed directly without delay on waiting for a review,
saving also time from the other developers.
Taking time to express my thanks for all his contributions so far and
reviving those unit tests, and I am looking forward to future collaboration!
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
Hello,
I'm trying to load data from a mysql table into a htable using a mysql
view. But when i start kamailio i get the following error:
ERROR: htable [ht_db.c:345]: ht_db_load_table(): Wrong db type [3] for
key_type column
Here a example of a view i'm trying to use:
CREATE OR REPLACE VIEW `view_blacklist_prefix` AS SELECT
prefix AS key_name, '0' AS key_type, '0' AS value_type, '1' AS
key_value, '0' AS expires
FROM openser.blacklist_prefix
GO
Should it be possible to use a mysql view as input for a htable?
Thanks,
Jan Hazenberg
Hi
Thanks in advanced if anyone can help me on this.
I have kamailio -4.2.3-1.1 running in front of asterisk 13.13.1. I'm using
Path support with asterisk so that I can forward Sip requests via kamailio
to the asterisk server running on a LAN IP. Kamailio is configured with a
WAN and LAN IP.
All works well I can register a sip endpoint to the asterisk server via
kamailio and make an receive calls, SIP OPTIONs from the asterisk server are
received and relayed by kamailio to endpoints. But I'm having issues when a
call is hung up asterisk appears to be trying to send the BYE via its own IP
address rather that the IP of the LAN interface on the kamailio server.
Should I be looking at the issue being on asterisk as it is not sending the
BYE via correct IP?
Y.Y.Y.Y = WAN IP of client endpoint
192.10.10.216 = LAN IP of asterisk
Example of BYE from sip debug on asterisk
--- (10 headers 0 lines) ---
Really destroying SIP dialog
'5d3ae6944a08b42c2078e237141e0e40@192.10.10.216:5060' Method: OPTIONS
-- <SIP/1001-00000011> Playing 'digits/17.ulaw' (language 'en')
-- Executing [*250@cos-all:8] Wait("SIP/1001-00000011", "1") in new
stack
-- Executing [*250@cos-all:9] SayUnixTime("SIP/1001-00000011", ",,IMp")
in new stack
-- <SIP/1001-00000011> Playing 'digits/4.ulaw' (language 'en')
-- <SIP/1001-00000011> Playing 'digits/oh.ulaw' (language 'en')
-- <SIP/1001-00000011> Playing 'digits/4.ulaw' (language 'en')
-- <SIP/1001-00000011> Playing 'digits/p-m.ulaw' (language 'en')
-- Executing [*250@cos-all:10] Hangup("SIP/1001-00000011", "") in new
stack
== Spawn extension (cos-all, *250, 10) exited non-zero on
'SIP/1001-00000011'
Scheduling destruction of SIP dialog
'OGRjYzJlYmE5MDhjYTdkMTM1ZTAxNTk2YWYwNDg5MTA.' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:1001@Y.Y.Y.Y:16078;transport=UDP> for
address/port to send to
set_destination: set destination to Y.Y.Y.Y:16078
Reliably Transmitting (no NAT) to Y.Y.Y.Y:16078:
BYE sip:1001@Y.Y.Y.Y:16078;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.10.10.216:5060;branch=z9hG4bK46ed8549
Max-Forwards: 70
From: <sip:*250@sip.domain.ie;transport=UDP>;tag=as60c5965b
To: <sip:1001@sip.domain.ie;transport=UDP>;tag=471c3a31
Call-ID: OGRjYzJlYmE5MDhjYTdkMTM1ZTAxNTk2YWYwNDg5MTA.
CSeq: 102 BYE
User-Agent: OMBUTEL PBX
Proxy-Authorization: Digest username="1001", realm="asterisk",
algorithm=MD5, uri="sip:sip.domain.ie", nonce="4aaa41f3",
response="136fac72a799f9507a7466b578786110"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
KAMAILIO LAN IP = 192.10.10.202
SIP status of peer on asterisk
* Name : 1001
Description : Infinity-Test
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : cos-all
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language : en
Tonezone : uk
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest : ringback
Mailbox : 1001@ombutel-voicemail
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "Infinity-Test" <1001>
MaxCallBR : 384 kbps
Expire : 3228
Insecure : no
Force rport : No
Symmetric RTP: No
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : Yes
Path support : Yes
Path : <sip:192.10.10.202;lr;received=sip:Y.Y.Y.Y:16078>
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.200.122:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1001
SIP Options : norefersub replaces replace timer
Codecs : (ulaw|alaw|g729)
Auto-Framing : No
Status : OK (19 ms)
Useragent : Zoiper for Windows 2.43 r24984
Reg. Contact :
sip:1001@192.168.200.122:5060;rinstance=3798b50d10d60d84;transport=UDP
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Gerry Kernan
Infinity IT | 17 The Mall | Beacon Court | Sandyford |
Dublin D18 E3C8 | Ireland
Tel: +353 - (0)1 - 293 0090 | E-Mail:
<mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie
Managed IT Services Infinity IT - <http://www.infinityit.ie/>
www.infinityit.ie
IP Telephony Asterisk Consulting -
<http://www.asteriskconsulting.com> www.asteriskconsulting.com
Contact Centre Total Interact -
<http://www.totalinteract.com> www.totalinteract.com
hi all, did someone can help me for with some configuration im a litle lost with inbount trafic with kamailio
i need to redirect inbound conection from itsp (over wan) to an internal PBX but it seem nat is not working (I do the test with jitsi on my desktop and sip connection seem initiate (200 ok) but no sound )
like contact information is not update on sip header
can someone help me a bit
MARTIN LAPOINTE
Spécialiste en infrastructure technologique
T (418) 543-2600 x2119
F (418) 543-0363
mlapointe(a)inso.ca<mailto:mlapointe@inso.ca>
www.inso.ca<http://www.inso.ca/>
365 Racine Est #204
Chicoutimi, QC
G7H 1S8
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Hello,
all the mailing lists related to the Kamailio project are using now
lists.kamailio.org as primary domain, including sr-dev and sr-users.
>From now on, the emails should be addressed to:
- sr-dev(a)lists.kamailio.org
- sr-users(a)lists.kamailio.org
The previous domain (lists.sip-router.org) should still work, so
existing discussions can go on as usual.
If anyone is encountering any problems or notices some information on
kamailio.org that needs to be updated, do not hesitate to reply here or
contact me.
In addition, URLs should use now https, browsing the archive with plain
http is redirected to https urls.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
Hi Guys,
Is it possible to disable using of dns in kamailio?
The thing is that not all INVITES that arrives to the proxy comes with ip
172.16.205.25/PUB_IP in the ruri. Some gateways sends a string in ruri ->
@myproxysip and others sends @other.domain.com
Request-Line: INVITE sip:1111111@myproxysip:5060;user=phone SIP/2.0
Request-Line: INVITE sip:1111111@other.domain.com:5060;user=phone SIP/2.0
Sometimes, myproxysip is from a public ip and sometimes is from a private
ip. So, it's difficult to know to which of kamailio's interfaces (pub or
priv) is.
ERROR: <core> [resolve.c:1694]: sip_hostport2su(): could not resolve hostn
ame.
Reconfigure those gw it's not possible. There are a lot of them and a lot
of people that won't like that idea.
Is possible to disable it? or What do you recommend me to do? Maybe using
/etc/hosts?
Any suggestion will be very appreciated.
Diego.