sr-users January 2017

sr-users@lists.kamailio.org

87 participants
114 discussions

NAPTR/SRV lookups fail when use_dns_failover is enabled.
by Jeremy Gadd 23 Jan '17

23 Jan '17

Hello, everyone! I'm trying to enable dns_srv_lb but I'm stuck just before I can get there. If I don't enable use_dns_failover, lookups work and calls route but always to the first host. We need to load balance, so that doesn't work very well. If I enable use_dns_failover, the lookups stop working completely and we get a dns lookup failure like so: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve " redacted.hostname.here.com" :unresolvable A or AAAA request (-7) This works: use_dns_cache=on dns_use_search_list=no dns_try_naptr=on # use_dns_failover=on This breaks it: use_dns_cache=on dns_use_search_list=no dns_try_naptr=on use_dns_failover=on I have checked to see that use_dns_failover is compiled in: version: kamailio 4.4.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2

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"Named Telephone Event" (I think) problem
by Pete Kelly 23 Jan '17

23 Jan '17

Hi I am running Kamailio 4.4 and have come across a strange SBC which is sending DTMF within the SDP body of a NOTIFY with a Content-Type audio/telephone-event. Has anybody come across this before? The content itself comes through on a tcpdump/ngrep trace as four dots "...." and Kamailio puts an empty string into the logs if I try to log the contents of $rb, however Wireshark manages to decode it as something readable "\005\000\a\320" and usable. (essentially the digit 5) Is there any way I can get access to this decoded version within Kamailio at all? Even if I have to pass it out to an external program? Many thanks for any help anyone can offer! Pete

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Best way to log incoming/outgoing sip messages?
by Pranathi Venkatayogi 23 Jan '17

23 Jan '17

Hi, What is the best way to get a complete dump of incoming/outgoing SIP messages of Kamailio server? I encountered “SIPTrace” module, but it writes to database. Is there a way we can write to syslog instead? Thanks Pranathi

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Kamailio IMS Secure Websocket closes connection
by Serhat Guler 22 Jan '17

22 Jan '17

Dear all, I am trying to make a call between 2 sipml5 client over WSS. Everything works fine and as expected when using WS. While using WSS, the clients are registered successfully, but when I make a call, the called party is disconnected from the network and the caller gets "User offline" message. There a wireshark capture here: https://drive.google.com/file/d/0B6LMw8kmoAMYMGhrYXNUVXJySEk/view?usp=shari… The call is made from Alice to Bob, the port Nrs: 4060: PCSCF, 5060:ICSCF, and 6060 SCSCF. WWS is running on 4443. The error message that is shown in Firefox console: "The connection to wss://192.168.0.11:4443/ was interrupted while the page was loading." Thanks in advance, Serhat

1 0

Get ip from contact on redirect
by Diego Nadares 20 Jan '17

20 Jan '17

Hi Guys, Anybody knows how to get the ip of the new contact of the redirect? I need it to update one xavp variable that contains the INVITE destination. 6 2017-01-20 12:44:55.138473 172.16.213.38 *172.16.208.121* SIP/SDP 85 Request: INVITE sip:01152780776@172.16.208.121:5060 | 9 2017-01-20 12:44:55.142296 172.16.208.121 172.16.213.38 SIP 661 Status: 302 Moved temporarily | 10 2017-01-20 12:44:55.156268 172.16.213.38 172.16.208.121 SIP 423 Request: ACK sip:01152780776@172.16.208.121:5060 | 16 2017-01-20 12:44:55.293349 172.16.213.38 *172.16.208.111* SIP/SDP 95 Request: INVITE sip:1152780776@172.16.208.111:5060;user=phone | My cfg: # Manage failure routing cases failure_route[MANAGE_FAILURE] { .... if (t_check_status("302")) { xlog("Got a 302 - redirecting"); get_redirects("*:1"); * $xavp(ip_b[$avp(current_route)]=>name) = "172.16.208.111";#Hardcoded for testing* xlog("Got a 302 - redirecting $hdr(Contact)"); route(RELAY); #t_relay(); } Thanks in advance. Diego.

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Siremis Multi domain
by Annus Fictus 20 Jan '17

20 Jan '17

Hi, I think Siremis not work correctly in a multi domain scenario if the extension number are the same for two domains. I explain a little more. I have two extension: 1 200(a)sip1.domain.org 2 200(a)sip2.domain.org On the Siremis I create the user 200 with the SIPuser role. When I login with this user, and go to subscriber data i can see bots extensions. I think is relate to this select on kamailio database: .....SELECT T0.id, T0.username, T0.domain, T0.password, T0.email_address, T0.ha1, T0.ha1b, T0.rpid FROM `subscriber` T0 WHERE T0.username='200' LIMIT 10 Any hint? Regards

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UAC authentication
by anfecora 20 Jan '17

20 Jan '17

can someone please guide me where to find the usage of auc module, thanks in advance. I am registering the uac in one end then i want to forward that registration to an external registrar, if i use $du some domain, some of the ipv6 info is going to be forwarded, i need to make it look more like an a b2bua than a outbound proxy therefore i need to use the auc module(also i need to cover all ipv6 info since i am proxying from ipv4 to ipv6). i am able to register the client to kamailio, then kamailio send the registration to the external registrar, but when the registrar sends back the challenge uac does not reply, any ideas how can achieve this. uac(linphone etc) --registration--->kamailio(registrar module)------register(uac)--->SBC sbc----invite---->kamailio(lookup)---------invite---->uac(linphone etc) this is the code i am using. # Handle SIP registrations route[REGISTRAR] { xlog("REGISTERING "); if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if(af==INET6) setbflag(FLB_IPV6); xlog("IT IS IPV6"); if (!save("location")) sl_reply_error(); route(REGSBCFWD); exit; } } route[REGSBCFWD]{ xlog("CREATING OTHER LEG TO THE SBC"); if(!is_method("REGISTER")) { return; } xlog("CREATING PACKAGE $si -- $fU-- $au--$du-- $tU --$ru"); $var(rip) = "sip.somedomain.com"; $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + "5060"; $uac_req(furi)="sip:" + $tU + "@" + $var(rip); $uac_req(turi)="sip:" + $tU + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $tU + "@" + "10.0.0.222" + ":" "5060" + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } REGISTER sip:sip.somedomain.com:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.222:5060 ;branch=z9hG4bK285f.13a7f0c0000000000000000000000000.0. To: <sip:14052423@sip.somedomain.com>. From: <sip:14052423@sip.somedomain.com >;tag=a82eaf71666096790d1397845fbd3254-ebaa. CSeq: 10 REGISTER. Call-ID: 357e21ae51d8416d-26605(a)10.0.0.222. Max-Forwards: 70. Content-Length: 0. User-Agent: kamailio (4.4.4 (x86_64/linux)). Contact: <sip:140524@10.0.0.222:5060>. Expires: 3600. . SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.0.0.222:5060 ;branch=z9hG4bK4b53.ad842757000000000000000000000000.0. To: <sip:14052423@sip.somedomain.com>. From: <sip:14052423@sip.somedomain.com >;tag=a82eaf71666096790d1397845fbd3254-37da. Call-ID: 357e21ae51d8416d-26603(a)10.0.0.222. CSeq: 10 REGISTER. WWW-Authenticate: Digest realm="sip.somedomain.com", nonce="f5831435-3e69-4b78-ab6d-dcb0897e477b", algorithm=MD5, qop="auth". User-Agent: somedomain.com Host:some proxy. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Content-Length: 0. .

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Return body/string from Perl library
by David Cunningham 20 Jan '17

20 Jan '17

Hello, We are using Kamailio's perl module and want our custom library to set the body on a SIP message. Is it possible to set a SIP message body from within Perl, or can we return a string to the Kamailio configuration to then use with set_body()? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019

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Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 19 Jan '17

19 Jan '17

root@homer02:~# netstat -altp Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 0 0 homer02.xxxxx.xxx:sip *:* LISTEN 16018/kamailio tcp 0 0 homer02.xxxxx.xxx:5066 *:* LISTEN 16018/kamailio Thanks. JR > If you use netstat, what is in the recv queue for tcp packets on sip ports? > > netstat -altp > > Cheers, > Daniel > > > On 18/01/2017 16:43, JR Richardson wrote: >> Yes, this is a sipcapture node. I'm listening on a switch port that is >> set to mirror my VoIP traffic. I see all SIP UDP/TCP packets on the >> mirror port and the Ethernet port of the host node. Just don't see any >> TCP packets process in kamailio, debug 3. UDP packets are processed as >> expected. >> >> My config is using port mirror for the capture parameters: >> >> modparam("sipcapture", "capture_on", 1) >> modparam("sipcapture", "hep_capture_on", 0) >> modparam("sipcapture", "raw_ipip_capture_on", 0) >> modparam("sipcapture", "raw_moni_capture_on", 1) >> >> modparam("sipcapture", "raw_sock_children", 4) >> modparam("sipcapture", "raw_interface", "eth1") >> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") >> modparam("sipcapture", "promiscious_on", 1) >> modparam("sipcapture", "raw_moni_bpf_on", 1) >> >> Is there a method I could diagnose if the SIP TCP Packets are getting >> from the kernel network process and the kamailio process? >> >> # ngrep -d eth1 -W byline host x.x.x.x | /var/run/kamailio/kamailio.pid >> >> Or pipe to kamailio local unix socket? >> >> I don't know, I'm just guessing. >> >> Thanks. >> >> JR >> >> >>> Somehow is not clear for me how you have the configuration there ... >>> before commenting further, this needs to be clarified. >>> >>> The node you presented the config is a sipcapture instance, right? What >>> is sending traffic to it? Is another kamailio with siptrace module? Or >>> the sipcature agent? Or you have a port mirroring in the router? >>> >>> Cheers, >>> Daniel >>> >>> >>> On 17/01/2017 16:37, JR Richardson wrote: >>>>> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >>>>>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >>>>>> using raw socket capture, I think this may be a sipcapture issue, >>>>>> debuging kamailio shows normal startup and processing of UDP SIP >>>>>> packets, but does not show any activity with TCP packets. >>>>> I never used HOMER sofar but when I saw your first message my thoughts >>>>> was that this can't work in a simple way since for TCP you need to >>>>> complete a 4 way handshake before you can start to send data. >>>>> >>>> Interesting. Are you referring to handshaking on the network stack or >>>> SIP TCP TLS handshaking? I guess I can see it two ways. >>>> >>>> 1) if your talking about TCP/IP handshake, even though the SIP packet >>>> comes into the mirror port on the host node, the kernel processing the >>>> TCP packet is not establishing a valid connection due to no TCP >>>> handshake because its only a monitor port, no transmit back, then the >>>> kernel network stack does not pass the SIP TCP packet to the kamailio >>>> process for capture because it drops the packet due to no valid >>>> handshake? >>>> >>>> 2) the kernel network stack is passing the SIP TCP packet to the >>>> kamailio process, but since kamailio cannot handshake back it drops >>>> the packet and does not process through the sipcapture module. This >>>> kinda breaks the whole capture ability for homer with SIP TCP. Using >>>> ngrep, I see all SIP TCP packets, invite -->, trying <--, session >>>> progress <--, request timeout <--, ack -->, etc... >>>> >>>> So how would I diagnose if the network stack is the culprit? Debugging >>>> kamailio is pretty straight forward, setup and listening for SIP TCP, >>>> but never see any processing of any TCP packets. >>>> >>>> Thanks. >>>> >>>> JR >

2 1

Phone number formats per country
by Daniel-Constantin Mierla 19 Jan '17

19 Jan '17

Hello, slightly off-topic, but related to voip -- does anyone have a link that can quickly share to a site with up to date details about phone numbering plans for as many country as possible? Ideally to include the toll and premium number prefixes, the split between mobile and fixed lines if it is the case. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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sr-users January 2017