sr-users February 2016

sr-users@lists.kamailio.org

80 participants
133 discussions

want to go mobile, common issues ?
by Sébastien Brice 27 Feb '16

27 Feb '16

Hi everyone. I have enjoyed talk about realtime of last FOSDEM, talk about K. was good. I want to know how do you solve sip dialogs with mobile? actually I know REGISTER is important to authenticate and this request generally last 1hour until then you have to register again. How to solve that for mobile telephony (and what about idle) ? do cellular phone use this schema ? you have to go UDP ? does TCP comes better to handle mobile telephony? I want to ask about IMS as well :-) I heard many good things about Image Multimedia Systems, but in concrete terms what is it? Does IMS target mobile only ? i heard about Boghe, it has many more parameters regarding DNS SRV and extra options. What's so different from my traditionnal Linphone ? thx you

2 1

ims_qos module support IPv6 address?
by Ayon Pal 26 Feb '16

26 Feb '16

Hi, I have been recently configuring kamailio IMS (v 4.3.4) for IPv6 user registration. I have my PCSCF Gm interface, ICSCF Mw interface and SCSF Mw interface all listening on IPv6 address on port 5060. PCSCF Rx interface is configured with IPv4 address to communicate with PCRF. Similarly ICSCF and SCSCF are also configured with IPv4 address to communicate with HSS (not Fraunhofer HSS) . My Linphone client configured with IPv6 address is sending register message to PCSCF. PCSCF throws the following error on while sending the AAR to PCRF *ubuntu kamailio[22627]: INFO: <script>: REGISTER (sip:001010000000004@TestNetwork (5555:0:0:1:ABCD:0:0:1:9060) to sip:001010000000004@TestNetwork, KFEtBsO-dm)ubuntu kamailio[22627]: ERROR: ims_qos [rx_aar.c:878]: rx_send_aar_register(): Unable to add framed IP AVPubuntu kamailio[22627]: ERROR: ims_qos [rx_aar.c:904]: rx_send_aar_register(): unexpected errorubuntu kamailio[22627]: ERROR: ims_qos [mod.c:1145]: w_rx_aar_register(): Failed to send AARubuntu kamailio[22627]: ERROR: ims_qos [mod.c:1189]: w_rx_aar_register(): Error trying to send AARubuntu kamailio[22627]: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction* On debugging, I found that the ip version is hard coded to AF_INET.( a comment is also written TODO: IPV6 support) in ims_qos/mod.c file. On changing it to AF_INET6 ( I did this 'cos I saw IPv6 based checking on other files like "ims_qos/rx_avp.c") I can see AAR from PCSCF to PCRF, but Framed IPv6 Prefix AVP is incorrect. So, I would like to know if ims_qos module supports IPv6 address ? Regards, Ayon

1 0

Kamailio: Internal call
by Vicente Falco 26 Feb '16

26 Feb '16

I have a kamailo with 2 ethernet cards, the first with ip public and the other for asterisk virtual machines. When I call to a phone number (not intenal) it works, but whe I call to a asterisk extension... it does'nt work. (message: service unavailable 500) I have the kamailio with multidomain. What can I do? -- Un Saludo, Vicente Falcó

2 1

Media server can't return ACK correctly
by Alexandru Covalschi 26 Feb '16

26 Feb '16

Hello everyone. The setup is: Carrier ip is CARRIER_IP Public network Kamailio IP will be PUBLIC_IP Private network Kamailio IP will be KAMAILIO_PRIVATE_IP Private network Freeswitch IP is FREESWITCH_PRIVATE_IP ACK CARRIER_IP -> PUBLIC_IP->FREESWITCH_PRIVATE_IP And freeswitch tries to actually send ACK back to PUBLIC_IP which he can't access. Kamailio trace: http://pastebin.com/raw/1W1sXuUa Freeswitch trace: http://pastebin.com/raw/KkZCwTTJ request_route: http://pastebin.com/raw/Y17pXUGY NATMANAGE route: http://pastebin.com/raw/0BpPDjN0 WITHINDLG route: http://pastebin.com/raw/5LpwSigF I'm seeking help with that - what parameter I need to change/add to solve that? Maybe it's a networking problem - but why then ACK reaches Freeswitch and all other requests flow OK? Thanks in advance, Alex -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 9

[4.2.6] Keep-alive sent by dialog module seems does not refresh the dialog...
by Jack Wang 26 Feb '16

26 Feb '16

Hello, I used the below settings to test the keep-alive function within a dialog: modparam("dialog", "default_timeout", 60) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) and use dlg_set_property("ka-src") or dlg_set_property("ka-dst") to send OPTIONS to caller or callee. A few questions to ask: (1) What's the difference between ka_timer and ka_interval? I read the http://www.kamailio.org/docs/modules/4.2.x/modules/dialog.html , but still can not understand very clearly. I know the ka_interval means OPTIONS will be sent after 30 seconds when dialog started or last keep-alive has been done. but I don't know how ka_timer works. (2) From the packets captured I can see server sent OPTIONS and the client (caller or callee) sent back 200 OK. According to the document, "The dialog timeout is reset each time a sequential request is processed" However, the dialog was still timeout, it seems that the timeout value was not reset. And I make the clients send the OPTIONS and frequently , the dialog never expires now. So the refresh only works on the keep-alive sent by clients not the server here? Thanks :)

1 0

Binary interface: usrloc replications - restore registrations after node restart
by evoip＠ukr.net 25 Feb '16

25 Feb '16

Hi ! It's really nice option - that now Opensips nodes can replicate "SIP registrations" and "Dialogs" between each other w *"bin_listen" options. http://www.opensips.org/Documentation/Interface-Binary-2-1 * I have a few question related to cluster architecture with fail over: 1). - If one of the "Opensips phone edges" was restarted - can it catch all USRLOC registrations throught binary interface from alive nodes ? - if it can't do restoring of the database through binary interface - maybe you have an idea how to do that ? 2). - it's share dialogs information - if one node is suddenly down, can call dialog be proceeded on other node ? With kind regards, Ewgeny Berladin Voip Engineer

2 1

how to drop 200ok and survive?
by Uri Shacked 25 Feb '16

25 Feb '16

Hi, for some strange reason, ask my regulator.... i need to manipulate certain calls. the scenario goes like this: 1. caller sends invite to kamailio. 2. kamailio transfer the call to asterisk. 3. asterisk send progress and play "hello". 4. asterisk creates a new call (dial) to the same kamailio with destination callee. 5. the callee answers the call. here, i need to block the 200ok. so that the caller does not receive it. i managed to block it with t_suspend(). but, there is no bidirectional media. the 183 progress was sent with sendreceive. it seems the asterisk is waiting for the ACK in order to open both ways for media. i tried to use uac_send_req() but it is being sent with no to tag. and when i try manipulating the uac_req(turi) it does not help because it takes all the string i entered and wraps it with <>. any ideas? BR, Uri

3 5

Credential server
by Marek Moravčík 25 Feb '16

25 Feb '16

Hi guys, I am currently doing research in SIP security. As described in RFC 6072, there can be credential server storing public and private keys (certificates). I have question, if there is any support from Kamailio for this, or if somebody has tried similar things with Kamailio. Thanks! Best regards Marek Moravcik

2 2

Can not start kamailio 4.3 and 4.4. Fork errors with working configuration.
by Yuriy Gorlichenko 25 Feb '16

25 Feb '16

Checked config at another servers instanses. I tried to start 2 workable configs. All starts ends with 3 type of errors error.log error2.log error3.log Debug outputs of 3 times started one config on one same server. I think error outside kamailio because 4.3 and 4.4 also compiled successfully. But at this time I can not understand where is trouble. In first file I see that kamailio can not connect to database but user/pass is correct, and I also see that kamailio can not make fork of process. 2 and 3 process have different troubles but base is one - can not fork.

2 1

ovh trunk
by Sébastien Brice 24 Feb '16

24 Feb '16

Hi everyone I followed this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and got it working (101, 102 and 103) can call eachother. But now i am trying to figure Asterisk's role out. I am more an ipbx person and i am used to register providers trunk in asterisk/sip.conf file, like this: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension] doing that i got plenty of OPTIONS request and 200 OK reply between my Kamailio and my provider (and it is a bit noisy) doing that i feel missing kamailio's logic and power to deal with externals trunk provider The thing is i need my authenticated users (101,102,103) be capable dialing my trunk and requesting INVITE for non-local request. What is the best way to achieve that? My DID provider gave me user/passwd/realm. I heard about avp special variables (auth_XXXX_avp and uac) and some snippets config that could help me to go there. Is that efficient to place the routing's logic to Kamailio and how to do that with my ovh trunk? thx you

2 1

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sr-users February 2016