sr-users May 2011

sr-users@lists.kamailio.org

78 participants
129 discussions

htable ::size has wrong value
by Juha Heinanen 24 May '11

24 May '11

i have entries in htable with key name lpi::20033::prefix: # sip-proxy_ctl mi sht_dump htable | egrep 20033 lpi::20033::prefix[0]:: +3583796294 lpi::20033::prefix[1]:: +35837824030 lpi::20033::prefix::size:: 1 as you see in above, size of that key name is 1 although there clearly is two entries in the table with that key name. according to readme ::size is number of items: 3.7. array_size_suffix (str) The suffix to be added to store the number of items in an array. i verified with another key name where there was only one prefix, and then ::size was again 1. so sometimes ::size is correct and sometimes incorrect. this looks like a serious bug to me. any comments? -- juha

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Telefonia ip
by rom ignacio 24 May '11

24 May '11

friend I am interested in learning more about VoIP or IP telephony, I read a lot about you and I still have more to read, as it can not released through theofficial documentation page errors Please let me send them am email, thanks I DEVELOP voip server to sip in Venezuela, I need all the help possible. thanks *Rommel Malave .M ** * *T.S.U Universitario Extracción y Producción de Gas * * T.S.U en Informatica*** 04148241688

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Re: [SR-Users] app_lua in kamailio
by Daniel-Constantin Mierla 24 May '11

24 May '11

Hello, please do not write direct private emails for questions about kamailio, use public mailing list sr-users(a)lists.sip-router.org (cc-ed) -- private emails are ignored, after the first notification. exit is not a function exported by a module, it is implemented in core. On the other hand, you try to stop the kamailio cfg interpreter when lua interpreter is executing the Lua script, so it won't work straightforward. Set a variable in kamailio from Lua, then check it after executing the lua script to decide to further exit or not. Cheers, Daniel On 5/24/11 11:44 AM, Bret McDanel wrote: > I am having a terrible time trying to get a conditional response from a > lua script. > > Basically I want > > if error then > sr.sl.send_reply(404,"Not Found"); > sr.modf("exit"); -- this does not work! > end > > > How can I have such a conditional? I have tried with return values from > the script to see if that would change anything but that did not seem to > work, sr.modf("exit") is not found. > > Any assistance would be greatly appreciated. > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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struct run_act_ctx
by Bruno Bresciani 24 May '11

24 May '11

Hi, I'd like to know the functionality of the struct run_act_ctx. I need to adapt my route module to use this struct, the function do_action require this parameter. I'm migrating of the kamailio 1.5.0 to 3.1.2. Regards,

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Re: [SR-Users] 403 Proxy forbidden replay for invite that send sip gateway
by hala alramli 24 May '11

24 May '11

thanks for your reply Klaus ,attached my config file could you please take a look ,i found the line that send that replay but i do not know why .? could you help me i am stuck --- Mer 18/5/11, Klaus Darilion <klaus.mailinglists(a)pernau.at> ha scritto: Da: Klaus Darilion <klaus.mailinglists(a)pernau.at> Oggetto: Re: [SR-Users] 403 Proxy forbidden replay for invite that send sip gateway A: sr-users(a)lists.sip-router.org Data: Mercoledì 18 maggio 2011, 13:04 Check your config and verify where 403 is sent. You have something like sl_send_reply("403","Proxy forbidden") in your config. regards Klaus Am 18.05.2011 10:48, schrieb hala alramli: > Hello every body, > could you help me ? > > i bought Geographical DID numbers from gateway provider , to receive > calls from pstn > > my problem that when gateway send invite my proxy send 403 forbidden ,i > dont know why what is wrong ,any help please. > > this is a track for th gateway invite. > > my proxy ip:79.12.254.153 > gateway IP :77.239.128.10 > my geographical number :0698962030 > > > > > > INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: <sip:0698962030@10.39.6.207:5060;user=phone> > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 INVITE > > User-Agent: CS2000_NGSS/8.0 > > Remote-Party-ID: <sip:0289089500@10.39.1.16;user=phone>; party=calling; > privacy=off; screen=yes > > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK > > P-Asserted-Identity: <sip:0289089500@10.39.1.16;user=phone> > > Max-Forwards: 69 > > Contact: <sip:77.239.128.10:5060;transport=UDP> > > Supported: 100rel > > Content-Type: application/sdp > > Content-Length: 236 > > > > v=0 > > o=PVG 1305215180830 1305215180830 IN IP4 77.239.128.10 > > s=- > > p=+1 6135555555 > > c=IN IP4 77.239.128.10 > > t=0 0 > > m=audio 49264 RTP/AVP 18 8 0 13 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=fmtp:18 annexb=no > > > > > > > > SIP/2.0 403 Proxy forbidden > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 INVITE > > Server: Kamailio (1.4.4-notls (x86_64/linux)) > > Content-Length: 0 > > > > > > > > > > > > > > > > > > > > > > ACK sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Content-Length: 0 > > > > > > > INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: <sip:0698962030@10.39.6.207:5060;user=phone> > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 INVITE > > User-Agent: CS2000_NGSS/8.0 > > Remote-Party-ID: <sip:026683011@10.39.1.16;user=phone>; party=calling; > privacy=off; screen=yes > > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK > > P-Asserted-Identity: <sip:026683011@10.39.1.16;user=phone> > > Max-Forwards: 69 > > Contact: <sip:77.239.128.10:5060;transport=UDP> > > Supported: 100rel > > Content-Type: application/sdp > > Content-Length: 236 > > > > v=0 > > o=PVG 1305215458700 1305215458700 IN IP4 77.239.128.10 > > s=- > > p=+1 6135555555 > > c=IN IP4 77.239.128.10 > > t=0 0 > > m=audio 41892 RTP/AVP 18 8 0 13 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=fmtp:18 annexb=no > > > > > > > > SIP/2.0 403 Proxy forbidden > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 INVITE > > Server: Kamailio (1.4.4-notls (x86_64/linux)) > > Content-Length: 0 > > > > > > > > > > > > > > > > > > > > > > > > sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Content-Length: 0 > > > > > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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SEMS 1.4.0 release
by Stefan Sayer 24 May '11

24 May '11

We are happy to announce the availability of the SIP Express Media Server version 1.4.0. SEMS is a high performance media and application server for SIP based VoIP networks. This release features a powerful Session Border Controller (SBC) module. From completely transparent B2BUA to customized URI/From/To, strictly filtered (messages, headers, codecs) with RTP anchoring, Session Timer enforcement, prepaid and call timer, the SBC facilitates interconnect and core routing in a simple and secure way. Thanks to the new multihoming support, SEMS can now be employed at the border of the networks. This addition also allows to overcome the bottleneck of one NIC - giving the possibility to fully exploit SEMS' great performance. In the app development area, the DSM language has matured to become a viable candidate also for implementing complex application logic, thanks to language constructs like for, if and functions. SEMS can be downloaded in source from its ftp site at: ftp.iptel.org/pub/sems/1.4/1.4.0/ Packages for debian/ubuntu should appear shortly on the OBS page: https://build.opensuse.org/project/show?project=home:team-sems Contributors to this release: Stefan Sayer, Raphael Coeffic, Bogdan Pintea, Matthew Williams, Peter Lemenkov, Anton Zagorskiy, Robert Szokovacs, Jon Bonilla, Thomas Gelf, Erik Dekkers, Tobias Lindgren, David J Many thanks to the companies sponsoring SEMS development, especially FRAFOS GmbH and TelTech Systems Inc. "E stands for Express." * Apologies if you receive multiple copies of this message. *

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~ operator has no effect
by Juha Heinanen 24 May '11

24 May '11

according to core cookbook: ~ : bitwise NOT however, looks like it has no effect. i have $var(test) = ~2; xlog("L_INFO", "test is <$var(test)>\n"); and i get to syslog: May 17 18:10:40 sip /usr/sbin/sip-proxy[29280]: INFO: test is <2> have i misunderstood the operator or is there a bug? -- juha

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Re: [SR-Users] IPv6
by Daniel-Constantin Mierla 24 May '11

24 May '11

On 5/20/11 6:33 PM, Jon Farmer wrote: > On 20 May 2011 17:25, Daniel-Constantin Mierla<miconda(a)gmail.com> wrote: > >> the module is good, but not for this case -- the warning seems to be from >> startup, saying that no reverse dns could be done to the ipv6 listen address >> and it is harmless (done for the purpose of building aliases list). >> >> Do you get the warning at runtime? >> >> Normally, the ipv6 should be in between square brackets. >> > Adding the address in question to my hosts file temporarily takes the > warning away :) ok, let me know how is going with the testing of ipv6 -- the support is in core since 2002, but not many entered the challenge so far :-) Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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newbie INVITE routing logic help
by Skyler 23 May '11

23 May '11

Hi all, Need some newbie advice on how to write INVITE routing for my scenario. I currently have kamailio+asterisk-realtime integration working in the following way: Outbound calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> SIP/PSTN This works, though I'd prefer to use Kamailio + LCR and gateway failover as Asterisk doesn't do this well. Here's what I want to do: (for now, small steps) Outbound Calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> Kamailio(3.1.3)+(lcr) -> SIP/PSTN I know I have to set kamailio as a GW in asterisk instead of the PSTN GW's and add the PSTN GW's to kamailio instead, though I'm at a loss as to how to write the routing logic so that kamailio sends calls from asterisk to PSTN. route[LOCATION] { ...non-INVITE check ends here... if(route(FROMASTERISK)) { # coming from Asterisk --- Does this mean check kamailio local users first before continuing? if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } ## -- If match 11 digits and starts with '1' or if match 10 digits and starts with [2-9] if($rU=~"^(1\d{10})$") || if($rU=~"^([2-9]\d{9})$") { xlog("L_INFO", " FROMASTERISK: This will become route(NORTHAMERICA); \n"); exit; ## to be removed } else { xlog("L_INFO", " FROMASTERISK: This will become route(INTERNATIONAL); \n"); exit; ## to be removed } # new call - send to Asterisk route(TOASTERISK); } Will this logic work? I haven't determined how to write route(NORTHAMERICA) or route(INTERNATIONAL) yet so I'm just logging to debug the logic and syntax. Thanks in advance.

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Kamailio to handle call states
by Yeris Antonio Madrigal Castro 23 May '11

23 May '11

Hi I am planing to change my old T1-E1 pbx, wich is AVAYA, for an asterisk cluster. I am planning this for a call center and I spec 80 currents calls for 30 agents sometimes, so I need to handle big queues and call and agent states, is it possible for kamailio handle the call states and agent states, and leave asterisk for the call center functions? Thanks in advance

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sr-users May 2011