sr-users February 2016

sr-users@lists.kamailio.org

80 participants
135 discussions

Help
by Kevin Farrell-Pelletier 29 Feb '16

29 Feb '16

Hi, I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VMs (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some kind of SIP Router, which would authenticate clients and route their SIP traffic to the appropriate server. By doing so, we are hoping to further secure our infrastructure and to possibly have only one Public IP (which would resolve to the Private IP of the SIP router). The Asterisk servers serve IVR/RINGGROUPS/OUTBOUNDTRUNKS/INBOUNDROUTES/OUTBOUNDROUTES. The Sip Router would therefore route all SIP traffic between the phones and the Asterisk servers, ad the phones would register to the SIP Router. I have tried many solutions (Kamailio, OpenSER, siproxd, Brekeke), but have not been able to configure these services to work the way we want them to. I am including a chart along with this email to outline what we would like to accomplish. Any suggestions or guides would be immensely appreciated. Thank you all for your time. Kevin Farrell Pelletier - Technicien informatique TI // Réseautique // Téléphonie IP // Programmation IT // Networking // VoIP // Application development 9060, boul Parkway, Anjou, Québec, Canada H1J1N5 Téléphone / Phone : 514.907.2000 Ext.203 F : 1.888.582.4001 - SF/TF : 1.855.907.2001 Web : <http://www.connextek.ca/> www.connextek.ca Pensez ENVIRONNEMENT, c'est important: n'imprimez que si nécessaire Consider the ENVIRONMENT before printing

4 3

What is the meaning of "shmized"
by sgy 28 Feb '16

28 Feb '16

Hi, Sorry, but I just want to know the meaning of "shmized", I can't find it in the dictionary, thank you very much. typedef struct _cfg_group { int num; /*!< number of variables within the group */ cfg_mapping_t *mapping; /*!< describes the mapping betweeen the cfg variable definition and the memory block */ char *vars; /*!< pointer to the memory block where the values are stored -- used only before the config is shmized. */ -- Best Regards, Shengy

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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

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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/

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[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

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sr-users February 2016