sr-users September 2013

sr-users@lists.kamailio.org

89 participants
128 discussions

TEXTOPS parser problem with binary data in MIME body
by Klaus Feichtinger 17 Sep '13

17 Sep '13

Hello, I have a problem with Kamailio version 3.2.4 (tested also with 3.3.5) regarding "binary data" in message bodies. According RFC3261 it is explicitly allowed using binary data within MIME bodies: RFC 3261 section 7.4.1: SIP messages MAY contain binary bodies or body parts. When no explicit charset parameter is provided by the sender, media subtypes of the "text" type are defined to have a default charset value of "UTF-8". However, the Kamailio function "Regular Expression Transformation" (re.subst), which is based on the transformation class (exported by the textops module) is causing problems in our customer system. In regular scenarios, Kamailio is receiving SIP INVITE messages with a MIME body, which is containing following parts: - application/sdp (standard conform) - application/x-q931 (Cisco proprietary with BINARY data!) - application/gtd (Cisco proprietary with ASCII strings) The content of the "x-q931" and "gtd" body parts is depending on (UUS1) data that were received on the ISDN line. Kamailio has to forward User-User specific information from the ISDN line to SIP user agents (in form of the "User-to-User" header field). The content of these UUS1 data may contain also byte values "00", which is legitimate. In general, Kamailio is in every INVITE message searching specific content in the last body (application/gtd) and copying this content to a config variable. As soon as the x-q931 body contains nul values (0x00 in binary format), the parser stops at this position and cannot parse the rest of the message. Therefore, I am missing the information that should be copied to the SIP header field, as the parser stops before the end of the message body! As long as the message body does not contain 0x00, it is working fine! My question is: - is this a bug in Kamailio parsing functions? - is this a design issue of Kamailio text parsers (as binary data are allowed acc. RFC3261) - does anybody know a solution for this problem? This "bug" is causing big troubles.... Thanks in advance, Klaus Feichtinger P.S. trace data (1)...(3) of my problem (1) exemplary content of the x-q931 body ======================Sending: Binary Message Body Sep 13 10:50:19: Content-Type: application/x-q931 08 01 B4 05 04 03 80 90 A2 18 01 89 1E 02 82 88 6C 05 B1 35 30 30 34 70 05 B1 35 30 30 31 7D 02 91 81 7E 18 04 1D 42 75 25 92 43 31 62 94 00 00 2C 68 20 64 00 62 F2 10 41 B9 D7 BD 0D 0A (2) SIP INVITE message (debugger view) =============================INVITE sip:115101@<ipv4>:5060 SIP/2.0 Via: SIP/2.0/UDP <ipv4>:5060;branch=z9hG4bK171164E From: <sip:1101015004@<ipv4>>;tag=29E16410-2541 To: <sip:115101@<ipv4>> Call-ID: 1DBCDACB-1B9911E3-8992FF70-D2BED293@<ipv4> Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 90 CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:1101015004@<ipv4>:5060> Expires: 1800 P-Asserted-Identity: <sip:1101015004@<ipv4>> Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 820 --uniqueBoundary Content-Type: application/sdp Content-Disposition: session;handling=required v=0 o=CiscoSystemsSIP-GW-UserAgent 1770 5294 IN IP4 <ipv4> s=SIP Call c=IN IP4 <ipv4> t=0 0 m=audio 16384 RTP/AVP 8 0 c=IN IP4 <ipv4> a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 --uniqueBoundary Content-Type: application/x-q931 Content-Disposition: signal;handling=optional Content-Length: 62 4" l15004p15001}~Bu%C1b..,h d.brA9W --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,08,,1,5001 CGN,08,,1,,,5004 UUS,3,1d427525924331629400002c6820640062f21041b9d7bd CPC,09 FCI,,,,,,,y, GCI,185ef1291b9911e381d500270dff3fa0 --uniqueBoundary-- (3) config excerpt ============if (has_body("multipart/mixed")) { if (filter_body("application/sdp")) { remove_hf("Mime-Version"); remove_hf("Content-Type"); append_hf("Content-Type: application/sdp\r\n"); } else { xlog("L_WARN", " <route> message body part 'application/sdp' not found\n"); } $var(UUS)=$(rb{re.subst,/^(.*)UUS,.,([a-z0-9,]*)..[A-Z][A-Z][A-Z],(.*)/\2/s}); append_hf("User-to-User: $var(UUS)\r\n", "CSeq"); if !(subst_body('/Content-Disposition..session.handling=required....//s')) { xlog("L_WARN", " <route> substituting Content-Disposition FAILED!!! \n"); } msg_apply_changes(); if (search_body("a=[a-z]+:.+[\r\n]{4}$")) { #!ifdef WITH_XLOGDEBUG xlog("L_INFO", " <route> double CRLF at the end of the message body detected - is reduced to one now. \n"); #!endif $var(sdp) = $(rb{s.striptail,2}); set_body("$var(sdp)", "application/sdp"); } }

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Registering "upstream" with the uac module, but with authentication
by Steve Davies 17 Sep '13

17 Sep '13

Hi, I'm making slow but steady progress with my Kamailio project. My next task: I need to "relay" a registration to an upstream service. IE when one or more devices registers against my local registration service, I need to initiate a corresponding registration upstream. If all my local devices unregister, all all the registrations lapse, I want to drop the upstream registration. There is an example a bit like this in the Asterisk integration example. However in that case the Asterisk doesn't challenge for authentication. My upstream does. I can't find anything in the uac module docs about authenticating. Is there any way to handle this. One thing I did try was to create "uacreg" as a view, and use that as the source of upstream registrations: CREATE VIEW `uacreg` AS select distinct `s`.`id` AS `id`,`l`.`username` AS `l_uuid`,`l`.`username` AS `l_username`,`s`.`domain` AS `l_domain`,`l`.`username` AS `r_username`,`s`.`domain` AS `r_domain`,`s`.`domain` AS `realm`,`l`.`username` AS `auth_username`,`s`.`password` AS `auth_password`,concat('sip:',`s`.`domain`,':5060') AS `auth_proxy`,timestampdiff(SECOND,now(),`l`.`expires`) AS `expires` from (`location` `l` join `subscriber` `s`) where (`l`.`username` = `s`.`username`) This kinda worked - but didn't quite track the registration status. Do you think this idea has promise? Maybe something similar that works using triggers rather than a straight view might work better? Thanks, Steve -- Steve Davies: Technical Director, Connection Telecom (Pty) Ltd

4 15

Calling users on another kamailio server
by Brian Wallen 17 Sep '13

17 Sep '13

I currently have two independent kamailio servers. I'd like to set them up in a way that user1 on server1 can make a call to user2 on server2. After searching I've come up with two ways that this might be able to be done. Can someone please sanity check these or point me in the right direction? 1. Have one registrar server and convert the other server to a proxy 2. Keep them both as registrars and somehow make them each aware of the users on the other server I like 2 better because if one server went down users on the other server would still be up. The only thing is I don't know how to set the servers up to communicate with each other.

4 15

kamailio presence and XCAP related issue
by Rupayan Dutta 17 Sep '13

17 Sep '13

Dear All, .May be I couldn't express my doubt properly ijn my earlier mails.Actually we were testing OfficeSIP server with SipML5 client for multiparty videoconferencing with presence enabled for showing buddy list and it is working properly.Now as we moved to kamailio with same client and configuring kamailio with Websocket and presence support, I have noticed that the presentity and xcap tables are not populated but location,watchers and active watchers tables are showing correct entries after a user logs in using sip over websocket from SIPML5. I have followed the steps as described in " http://kb.asipto.com/kamailio:presence:k31-made-simple". As Sipml5 sends PUBLISH/NOTIFY request in pidf format is this is the possible cause?As I am a person from java based web application technology it is really hard for me to detect the root cause.Please give me some idea regarding this issue.

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Asterisk and Kamailio Integration CAPACITY ISSUE
by Hassan Wajahat 16 Sep '13

16 Sep '13

Hi, I followed your tutorial and the setup is running seamlessly. But the question is that I read somewhere that asterisk can make up to 300 to 400 concurrent sip calls. Higher than that will cause problems. Will that issue appear here as well since you are using asterisk in the sip calling process?? Regards, Hassan Wajahat

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Kamailio realtime fails to forward register with Asterisk.
by imperium broadcast 15 Sep '13

15 Sep '13

Hi all I have been trying to figure out where Im going wrong with following this guide: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb The only thing that is different is I am using Asterisk 1.8.22.0 Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk. Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing. Realtime works fine when I register the phone on Asterisk. Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite. Am I missing something so obvious :-/ This is what I have in sip.conf "IP change to protect the innocent!" [kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all and this is the sip trace for the register U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000(a)127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: <sip:102@172.16.0.112:5060>. Expires: 120. . U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000(a)127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: <sip:102@172.16.0.112:5060>. Expires: 120. and I have this for my Asterisk binds in kamamilio.cfg #!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif -- Regards Mick

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issues with dispatcher rerouting to next server when getting a 500 from freeswitch
by Joseph Watson 14 Sep '13

14 Sep '13

Hello Everyone, I am really having an issue getting kamailio to try another dispatch route when I get a 500 back from one of my freeswitch servers. Currently I have a list of freeswitch servers located in my dispatch.list file. If one of then is to busy it will respond with a 500 Maximum Call in Progress. I am trying to get Kamailio to go to the next route in the dispatch.list when it get one of these 503 and if it should exhaust the whole list respond with a 503. Here is my kamailio config ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=4 log_stderror=yes #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=10 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:eth0:5060 sip_warning=no ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("xlog", "buf_size", 8192) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) modparam("tm", "remap_503_500", 0) # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") #modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") #modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") #modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #modparam("dispatcher", "ds_append_branch", 1) ####### Routing Logic ######## # main request routing logic route { append_hf("X-VSIP-CUST-IP: $si\r\n"); route(DISPATCH); route(RELAY); } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } # xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); return; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("DISPATCH"); route(RELAY); exit; } } } route[RELAY] { if (!t_relay()) { send_reply("503", "LB01 - All Circuits Are Busy"); # sl_reply_error(); } exit; } I am very new to kamailio and still learning my way around. Any help you can provide with be fantastic! Thank you, Joe Watson

2 1

json module / quotes in result
by Florian Lohoff 13 Sep '13

13 Sep '13

Hi, is it correct that the json_get_field returns the quotes from the json document? Example: { "foo": "bar" } json_get_field("$var(json)", "foo", "$var(result)"); result will contain "bar" with quotes ... I was a little astonished to see that. Flo -- Florian Lohoff f(a)zz.de

3 2

New Developer: Charles Chance
by Daniel-Constantin Mierla 13 Sep '13

13 Sep '13

Hello, I want to announce that a new person got developer GIT write access to repository: Charles Chance. He is for long time in the community, sending patches in the past to modules such as memcache. His immediate goal is to care of dmq module, having a set of patches to be committed as well as plans to integrate it within more modules. His git commit id is: cchance My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings at http://www.asipto.com -

3 2

NAT function
by Kethzer Docteur 13 Sep '13

13 Sep '13

I need help to fixed my NAT configuration I have nat module load but nat configuration is not working. ANY HELP PLease #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" loadmodule "nat_traversal.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@67.215.8.130") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } -- Kethzer Docteur

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sr-users September 2013