sr-users June 2013

sr-users@lists.kamailio.org

94 participants
170 discussions

simple way to access RR headers/uris and to remove them
by Klaus Darilion 18 Jun '13

18 Jun '13

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

3 7

Re: [SR-Users] REFER
by Daniel-Constantin Mierla 18 Jun '13

18 Jun '13

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.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

1 0

A general SIP question
by Gang Wang 18 Jun '13

18 Jun '13

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

2 1

e72 stops responding
by hiro 18 Jun '13

18 Jun '13

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

2 1

uac_redirect - logging/acc q value
by Geoffrey Mina 18 Jun '13

18 Jun '13

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

2 1

Dispatcher mode for Active non-probing mode
by Mino Haluz 18 Jun '13

18 Jun '13

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

2 1

regarding kamctl command to show online users
by Sunil Chandrasekharan 18 Jun '13

18 Jun '13

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?

2 1

Dispatcher module without OPTIONS keepalive pinging
by Efelin Novak 18 Jun '13

18 Jun '13

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

2 1

FW: Akhmad Dhany
by Akhmad Dhany 17 Jun '13

17 Jun '13

http://puresl.eemdevteam.com/zypn/nhq/dixo/vus/tyfacd/ubhp.html Akhmad Dhany dlv/div>

1 0

Kamailio and Asterisk integration
by Lucky Santiago 17 Jun '13

17 Jun '13

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

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sr-users June 2013