Hi!
Is there a simple way to:
- retrieve the RR headers (or Route/Contact ...) and to
- selectively remove them?
E.g. retrieve <2> and delete <3>
I do not want to care about the format, eg:
Record-Route: <1>, <2>, <3>
or
Record-Route: <1>
Record-Route: <2>
Record-Route: <3>
Thanks
Klaus
Hello,
On 6/15/13 12:01 PM, Tony Turner wrote:
>
> Hi,
>
> I currently have:-
>
> Kamailio -----Freeswitch--------NodeMax SIP/SS7------PSTN
>
> I have users on Kamailio registered 1000, 1001 to 1004 and make calls
> to each other and to PSTN and PSTN to users.
>
> Now I want to keep transfers off network so basically on Kamailio.
>
> So currently sip client 1000 calls 1001 it stays on Kamailio (that
> works), but if inbound from PSTN to 1000, 1000 cannot transfer to 1001
>
> The REFER is sent to freeswitch, can I get Kamailio to just transfer
> from 1000 to 1001.
>
> I don't see any reason why I would want a transfer to be sent to
> freeswitch and down to my Nodemax CTI SIP/SS7 gateway. It would seem
> sensible for the proxy to deal with a user to user transfer.
>
kamailio is a sip message router, will send the REFER based on R-URI and
Route headers. Based on SIP specs, REFER has to be handled in client
side, the proxy just delivers it.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
Hi,
I'm new to SIP server. I have a task on hand and try to research for a
solution. I need to develop a service emulator for handling RCS requests.
is It possible to use Kamailio SIP server to handle GSMA's RCS-Standards
compliant requests? Please point me to the right doc/tutorials.
Thanks in advance!
Gang
8 is a Nokia E72 calling in via kamailio. After the Invite I don't get
any other packets from the phone any more. It seems like the TCP
connection gets killed completely by something. It only starts working
again after I reboot the phone and it is completely reproducable.
14:06:51.457926 IP 82-171-55-218.ip.telfort.nl.10658 >
koln-4db447dd.pool.mediaWays.net.sip: Flags [P.], seq 2401:3615, ack
1416, win 65039, options [nop,nop,TS val 113865285 ecr 859201823],
length 12[1616/1930]
E....-@.<.w.R.7.M.G.)........D#g...........
..rE36a.INVITE sip:3@myhost;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 192.168.1.34:5060;branch=z9hG4bKggcacug4n6qn494dj0g2ear;rport
From: <sip:8@myhost>;tag=v8o9g80af9hc7j2304ao
To: <sip:3@myhost>
Contact: <sip:NQBMFpVZDLXb4Ju6v0ZZ@192.168.1.34:5060;transport=TCP>
Supported: 100rel,timer
CSeq: 5411 INVITE
Call-ID: fjao86anoIcFLw4lwjAR-waVvK4c4f
Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE
User-Agent: Nokia RM-530 091.004 (en)
Expires: 120
Privacy: None
Session-Expires: 1800
Max-Forwards: 70
Proxy-Authorization: xxx
Content-Type: application/sdp
Accept-Language: en
Content-Length: 413
v=0
o=8 63539388406935125 63539388406935125 IN IP4 192.168.1.34
s=-
c=IN IP4 192.168.1.34
t=0 0
m=audio 49152 RTP/AVP 100 0 8 97 98
a=sendrecv
a=rtcp:49153 IN IP4 192.168.1.34
a=rtpmap:100 AMR-WB/16000
a=ptime:20
a=maxptime:200
a=fmtp:100 mode-change-period=2; mode-change-neighbor=1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-15
14:06:51.458086 IP koln-4db447dd.pool.mediaWays.net.sip >
82-171-55-218.ip.telfort.nl.10658: Flags [.], ack 3615, win 3640,
options [nop,nop,TS val 859201829 ecr 113865285], length 0
E..4D%@.@..xM.G.R.7...)..D#g...P...85......
36a%..rE
14:06:51.523134 IP koln-4db447dd.pool.mediaWays.net.sip >
82-171-55-218.ip.telfort.nl.10658: Flags [P.], seq 1416:1767, ack
3615, win 3640, options [nop,nop,TS val 859201836 ecr 113865285],
length 351
E...D&@.@...M.G.R.7...)..D#g...P...8(d.....
36a,..rESIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TCP
192.168.1.34:5060;branch=z9hG4bKggcacug4n6qn494dj0g2ear;rport=10658;received=82.171.55.218
From: <sip:8@myhost>;tag=v8o9g80af9hc7j2304ao
To: <sip:3@myhost>
CSeq: 5411 INVITE
Call-ID: fjao86anoIcFLw4lwjAR-waVvK4c4f
Server: kamailio (4.0.1 (arm/linux))
Content-Length: 0
no response from phone...
I'm fishing in the dark, any ideas? Anything else I could check?
thanks
Greetings,
We are considering implementing an LCR which functions by returning a 302
with a list of our upstream carriers ordered by cost per minute.
Is there a way with the uac_redirect module in 1.5 to write the q value for
the branch that answers so we can calculate the cost of the call?
Thanks,
Geoff
Hi,
sorry, but from documentation I cannot understand what mode should I choose
for dispatcher host, so that it will not be probed at all and marked always
as active. I have sipp on another side, which does not respond to any
OPTIONS, so I want to temporarily disable checking.
Thanks,
Mino
Hi ,
When i type the command "kamctl ul show"
it shows that
Domain :: location table=512 records=1 max_slot=1.
When i keep registering new users, the tag "max_slot" still shows as 1.
What is tag " max_slot" means? What is it doing/showing?
Does it hinder from me registering more users?
Hi.
Is there a possibility to turn off the pinging in dispatcher module for
selected gateways? I want them to be active (to use them for dispatching)
but I don't want to monitor them because one of my gateways does not
support OPTIONS or INFO correctly.
Kind regards
Efelin
Hi,
I am planning to deploy a kamailio as a front-end of a currently in
production voip gateway which is running in asterisk (a2billing) and i
can't picture out in terms of configuration on kamailio how the
registration, dialplan on kamailio goes as asterisk does, if you can
please show me a sample config on how that stuff works to fully
communicate with my asterisk server, i have read and setup a working
kamailio installation but i wasn't able to establish a full
communication between kamailio and asterisk. Thanks in advance.
Lucky
