Dear All,
I am running Kamailio (4.0) and RTPproxy server (1.2.1) server in my
set-up. When i call from Client A to B , the RTP packets are proxying
trhrough RTPproxy service to other end client, but no flow from client B
to client A. With this calls Are unsuccessful. I traced a SIP capture
(please find in the attachment) and analysed it, as in there only one
client 's (A) IP address is changed to RTPproxy IP address in SDP body.
What can be do to resolve this issue ?
Please anybody help me
And also find the attachment for my kamailio config file, and suggest me or
point me about any misplacement of NAT traversal function call in the
script. Please consider this and help me out as i am newbie in scripting.
Regards,
Nandini
Hello,
I have a Kamailio-3.1.1 running with its DB on MySQL running on a seperate
machine. Siremis is installed for interaction with DB. In web interface of
Siremis, domain name of users is not shown. How can I show that ?? Like my
domain name is "defg" and user is "abc". It shows online users as
"abc@HostNameOfUser" instead of "abc(a)defg.com".
Secondly does Siremis read information from Kamailio server or directly from
DB server ??
Regards,
Aawaise
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Hi Everyone:
I am a newbie in kamailio. Basically I am looking for a way to
restrict the registration or login in kamailio per user to just one.
The experience I want to setup is to unregister an older register/login
to kamailio by a new login/register after the register command has already
passed the authentication leg.
Is there an example for this? Thanks.
--
"When I look at you I see two people, ther person you are
and the person you are supposed to be. Someday these two
people will meet. And when they do, they will achieve great things"
- Gene Hackman, The Replacements
Catch up on me and check my BLOG at
http://mrihaveanopiniononeverything.blogspot.com
To whom it may concern:
We are configuring a SIP platform with Kamailio and Freeswitch with this
setup:
UAC 1 ==> Kamailio ==> Freeswitch
UAC 2 <== Kamailio <== Freeswitch
We followed the instructions in
http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
We are using Kamailio on kamailio 4.0.3
and Freeswitch FreeSWITCH Version 1.5.6b
Everything is working fine with the default setting. But when we
configured TLS what we noticed some errors.
UAC configured with no TLS calling a UAC configured with TLS works ok.
UAC configured with TLS calling a UAC configured with TLS gets
disconnected 15-30 seconds after answering the call.
UAC configured with TLS calling a UAC configured with no TLS gets
disconnected 15-30 seconds after answering the call.
I cant seem to find any error in the logs or in the ngrep.
Can anyone assist me regarding this issue. Good day.
--
"When I look at you I see two people, ther person you are
and the person you are supposed to be. Someday these two
people will meet. And when they do, they will achieve great things"
- Gene Hackman, The Replacements
Catch up on me and check my BLOG at
http://mrihaveanopiniononeverything.blogspot.com
Hi
I've a problem of this type:
On the first INVITE is all ok, insert record route with cookie of the
dialog:
Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.1661>
The I recieive a 422 for the session timer too small .
I resend the new invite with correct expire but with a new cookie dialog
value
Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.2661>
Then the call is connected (Record-Route:
<sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.2661>).
The there is a reInvite for the session timer , the script enter in the
routine WITHINDLG . At this point the dialog seems to be lost.
The accounting is made with dialog variable and in the 2 records INVITE and
BYE , the fields corresponding to these values are missing.
In all the other calls without the reInvite for session timers, the dialog
is found and the variables are present.
I don't understood what is problem ? Thanks for the help
Hi Guys,
I'm doing some work, with Legacy Cisco phones, its a long story but I need to manipulate signaling due to firmware limitations.
I am looking to implement a workaround, whereby I need to modify an ACK, so that it sends the RURI is sent to port 5060, and the destination Port at the IP layer is unchanged, I can do this with static values as below;
sethostport("1.2.3.4:5060"); t_relay("1.2.3.4","38596");
In terms of doing this for multiple sites I need to use pseudo variables, so I wondered how this could be achieved, as t_relay or forward dont support dynamic values to this point, or do they?
Any comments would be great.
Many thanks
Jon
Hi im receiving the following INVITE on my system, you can see that the
Record-Route headers are all over the place, is there a way to group the
Record-Route headers together followed by the via headers grouped together?
INVITE sip:22000017@81.21.42.184;uniq=DBFF66459C85D6171B30016D2422B SIP/2.0
Record-Route: <sip:81.21.32.84;lr=on;ftag=as4c938f35;did=6da.a472>
Record-Route: <sip:81.21.38.34;lr=on;ftag=as4c938f35>
Via: SIP/2.0/UDP
81.21.32.84;branch=z9hG4bK6a96.dc3413ac122b4d4b8fd61bf19d95b1cf.0
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bK6a96.c81d2c2ad30db0820f05268ea67896f7.0
Via: SIP/2.0/UDP
81.21.38.5:5060;branch=z9hG4bKterm-477861-22030313-22000017-74007
Record-Route: <sip:81.21.38.5;lr=on;ftag=as4c938f35>
Record-Route: <sip:81.21.38.34;lr=on;ftag=as4c938f35;did=6da.0ff1>
Via: SIP/2.0/UDP
81.21.38.34;branch=z9hG4bK6a96.5d12e421943fbaec75df9a6a8091d336.0
Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7e570efe;rport=5060
Max-Forwards: 16
From: <sip:22030313@192.168.10.189>;tag=as4c938f35
To: <sip:22000017@81.21.38.34>
Contact: <sip:22030313@192.168.10.189:5060>
Call-ID: 74d7379b710c82f956d7cb31615c2581@192.168.10.189:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(11.7.0)
Date: Fri, 04 Apr 2014 11:33:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257
P-hint: outbound
P-hint: outbound
v=0
o=root 1396887739 1396887739 IN IP4 81.21.38.36
s=Asterisk PBX 11.7.0
c=IN IP4 81.21.38.36
t=0 0
m=audio 42746 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
thanks in advance!
Phillip
Hello all,
I would like to ask, how can I load new module in Kamailio 4.1.2? Actually,
I have an issue, when I tried to compile my kamaiio.cfg I've got error:
root@kamailio:/usr/local/# kamailio -c kamailio.cfg
loading modules under /usr/local/lib64/kamailio/modules/
0(25392) ERROR: <core> [sr_module.c:587]: load_module(): ERROR:
load_module: could not find module <websocket> in
</usr/local/lib64/kamailio/modules/>
0(25392) : <core> [cfg.y:3408]: yyerror_at(): parse error in config
file /usr/local/etc/kamailio/kamailio.cfg, line 323, column 12-25:
failed to load module
0(25392) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed
to find command ws_handle_handshake
0(25392) : <core> [cfg.y:3411]: yyerror_at(): parse error in config
file /usr/local/etc/kamailio/kamailio.cfg, line 1083, column 27:
unknown command, missing loadmodule?
ERROR: bad config file (2 errors)
when I look physically to the /usr/local/lib64/kamailio/modules/ there
is some modules, but websocket.so is missing. So, how can I get and
load module in Kamailio?
Thank you for help!
BR,
Patrik Kristel
Hello,
I experienced an issue yesterday.
I move a customer to Kamailio 4.0.4 (previously he was on an old SER
instance without issue).
He uses the same SIP account for connecting two IPBX (same platform, same
firmware).
When the second REGISTER came on Kamailio and, may be 500ms/1s after (I have
sufficient time to see 200 OK (2 bindings) on Wireshark), Kamailio crash
with coredump generated.
I know that I must analyse the coredump but I hope that could be a known
issue.
In addition, when I try to run "gdb" on the core files, I got the following:
Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m
256 -M 64'.
Program terminated with signal 11, Segmentation fault.
#0 0x00000030f2230f30 in escape_string_for_mysql () from
/usr/lib64/mysql/libmysqlclient.so.16
Missing separate debuginfos, use: debuginfo-install
glibc-2.12-1.80.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64
krb5-libs-1.9-33.el6.x86_64 libcom_err-1.41.12-12.el6.x86_64
libconfuse-2.6-2.el6.rf.x86_64 libselinux-2.0.94-5.3.el6.x86_64
mysql-libs-5.1.61-4.el6.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64
openssl-1.0.0-20.el6_2.5.x86_64 zlib-1.2.3-27.el6.x86_64
I guess that I'm missing packages to troubleshoot.
Many thanks.
Regards,
Igor.
