sr-users November 2013

sr-users@lists.kamailio.org

100 participants
160 discussions

407 proxy authentication required
by Joli Martinez 11 Nov '13

11 Nov '13

I am setting up Kamailio as an SBC and freeswitch to handle vm and conferences. When I dial 41 Kamailio sends the call to FS and I get the VM prompts, but when I dial 43 the call never hits FS. I am looking at FS in the cli and call never comes in. I get "407 proxy authentication required" from my sip client. I am new to Kamailio and cant seem to figure it out. My system is running CentOS 6.4. FS is installed via yum. Kamailio is complied running version 3.0 thanks,

1 0

implemented specifiers not processed
by Oliver Roth 11 Nov '13

11 Nov '13

Hi all Having some problems with the following part of the script in a timer: We have some kamailios running doing more or less the same job - they are used by a loadbalancer kamailio. Actually I wanted to check the ip or systemname of the current system do perform some actions depending on the used kamailio - based on a timer route[CDRS] { sql_query("ca","call kamailio_cdrs()","rb"); sql_query("ca","call kamailio_rating()","rb"); xlog("timer routine: time is %TF\n"); xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}>\n"); xlog("L_INFO", "CDRS- rated Host %Hi"); if %Hi = ipOfGateway1 then do something else if %Hi = ipOfGateway2 then do something different } What I get in log Nov 10 12:54:12 sipgw21 /usr/sbin/kamailio[16231]: ERROR: <script>: timer routine: time is %TF Nov 10 12:54:12 sipgw21 /usr/sbin/kamailio[16231]: ERROR: <script>: time [%Tf] method <%rm> r-uri <%ru> 2nd via <%{via[1]}> Nov 10 12:54:24 sipgw21 /usr/sbin/kamailio[16231]: INFO: <script>: CDRS- rated Host %Hi The %-variables are not processed ... what am I doing wrong?? Thanks Oli

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kamailio 4.0.4 Record-Route reply setting
by kamailio＠AaronLux.com 11 Nov '13

11 Nov '13

2 1

Carrierroute module
by Oliver Roth 11 Nov '13

11 Nov '13

Hi I get the following error when trying to load the carrierroute module load_module(): ERROR: load_module: could not open module </usr/lib64/kamailio/modules/carrierroute.so>: /usr/lib64/kamailio/modules/carrierroute.so: undefined symbol: cfg_set_error_function I installed Kamailio 4.0.3 by apt-get on Ubuntu 12.04 lts It seems, that with version 3.3 it is working ... root@sipgw22:/home/triotel# kamailio -V version: kamailio 4.0.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 17:03:55 Aug 19 2013 with gcc 4.6.3

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BUG? - Dialog module timeout BYE is not ok.
by Uri Shacked 11 Nov '13

11 Nov '13

Hi, Following this issue, and the issue "BYE dialog timeout bad syntax" from Julia. I have 2 kamailio servers. srv1 creates a dlg_timeout BYE and sends it to srv2 that forwards the BYE to the next sip server (some other server...). I can definitely see that the BYE that is being forward is malformed. The TO header info is not complete (the port section for example). When a "normal" bye is received from srv1 (not dlg_timeout) all works fine. One thing that i do in this scenario, on srv2 i use uac_replace() on the TO header. So, it is OK that the final destination is different (i do translations). I attached the BYE from both servers. Any ideas? Thanks, Uri You probably meant the To and From headers, I'm guessing. Yeah, as long as there is no display name component, URIs in headers like that don't have to go in <>s. The way Kamailio sends it is grammatically valid, per RFC 3261. The user agents are at fault for not understanding it correctly. Alex Balashov <abalashov at evaristesys.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: >*"BYE headers"? Which headers? *>* *>*Uri Shacked <ushacked at gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: *>* *>>*Hi, *>>* *>>*I think there is a bug with the BYE that is sent to the caller and *>>*callee *>>*when dialog timeout happends. *>>* *>> *The BYE headers are sent with no "<" or ">". *>>*So, some sip singaling points decline the BYE. *>>* *>>*BR, *>>*Uri *>>* *>>* *>>*------------------------------------------------------------------------ *>>* *>>*_______________________________________________ *>>*SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing *>*list *>>*sr-users at lists.sip-router.org<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> *>>*http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users *>* *>*-- *>*Sent from my mobile, and thus lacking in the refinement one might *>*expect from a fully-fledged keyboard. *>* *>*Alex Balashov - Principal *>*Evariste Systems LLC *>*235 E Ponce de Leon Ave *>*Suite 106 *>*Decatur, GA 30030 *>*United States *>*Tel: +1-678-954-0670 *>*Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ *>* *>*_______________________________________________ *>*SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list * >*sr-users at lists.sip-router.org<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> *>*http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users * -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: < http://lists.sip-router.org/pipermail/sr-users/attachments/20130422/0c75272…>

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Wrong "To" header in CANCEL, when "To" has been rewritten by uac_replace_to() in branch route
by Julia 10 Nov '13

10 Nov '13

Hello, I know it was already discussed, that not recommended to modify "To" header and no routing should ever be done on "To". But as long as there are carriers that do it, we need full support for this manipulation for all messages in the call flow. I found in list a few mails with the same problem of wrong header "To" in CANCEL after "To" rewriting. And one solution from Vitaliy Aleksandrov : http://lists.sip-router.org/pipermail/sr-dev/2012-October/016657.html "Kamailio builds "To" header from the received INVITE. Patch to fix it: --- ./kamailio-3.2.3/modules/tm/t_cancel.c 2012-08-22 10:39:21.000000000 +0300 +++ ./kamailio-3.2.3-fixed/modules/tm/t_cancel.c 2012-10-01 18:25:57.000000000 +0300 @@ -284,7 +284,7 @@ if (cfg_get(tm, tm_cfg, reparse_invite)) { /* build the CANCEL from the INVITE which was sent out */ cancel = build_local_reparse(t, branch, &len, CANCEL, CANCEL_LEN, - &t->to + NULL #ifdef CANCEL_REASON_SUPPORT , reason #endif /* CANCEL_REASON_SUPPORT */" We use header "To" manipulation in branch route (after failure route) and this patch solved our problem with "To" in CANCEL. But a patch is not committed. Is it possible to commit this solution? Thank you, Julia

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Modify/Edit INVITE message.
by Aziz 09 Nov '13

09 Nov '13

Hello. I have a question about Kamailio configuration. Can I edit or modify the INVITE message content? For example, remove "To" or "Via" fields in the INVITE content. If yes, how can I do? Thanks.

2 1

#TM: timing problem in serial forking (INVITE vs. CANCEL)
by Klaus Feichtinger 08 Nov '13

08 Nov '13

Hello list, I have troubles with serial forking in kamailio 3.2.4, which is mixed with parallel forking. In the scenario that an initial INVITE message, which is addressed to sip:A, is coming in to the server, it is doing a lookup in the DB and forking (parallel) the request to e.g. 3 SIP user agents. I have set the timer to 20 seconds transaction timeout and after that timeout, the server is handling the original request in the FAILURE_ROUTE[xy]. In this failure route, the request-URI username is overwritten to an alternative one -- e.g. sip:B. Then the server is doing a DB lookup again and forking the request to the number of registered user agents. A specialiy of this scenario is that it can be possible, that user agents have registered for username "A" and username "B" -- in other words: they are members of the parallel forking group in serial forking step 1 and step 2. When the CANCEL and INVITE message would be sent out (to the user agents) in the correct order, then it would be no problem. But in my case the server is sending the "new" INVITE message (2^nd step in the serial forking procedure) to user agents BEFORE the CANCEL request. Therefore, these user agents are rejecting the INVITE message with "500". Signalisation scenario ================== INVITE -> SRV SRV -> INVITE (branch-1) UA1 SRV -> INVITE (branch-2) UA2 SRV -> INVITE (branch-3) UA3 SRV <- 180 (branch-1) UA1 SRV <- 180 (branch-2) UA2 SRV <- 180 (branch-3) UA3 ..... [timeout] SRV -> CANCEL (branch-1) UA1 SRV -> CANCEL (branch-2) UA2 SRV -> INVITE (branch-4) UA4 SRV -> INVITE (branch-5) UA5 SRV -> INVITE (branch-6) UA3 (!!!) SRV -> CANCEL (branch-3) UA3 SRV <- 500 (branch-6) UA3 It was quasi reproduceable that only the last branch of the initial transaction had that timing problem (INVITE <- vs -> CANCEL). My question is: why does the server send the (last) CANCEL message only after the INVITE message for some branch(es)? Could this behaviour be prohibited in any way? Thanks in advance! Klaus

2 2

UUID generation
by Grant Bagdasarian 08 Nov '13

08 Nov '13

Hello, Any plans for building a module which generates a UUID or building it into the core? Daniel once told me to use $sruid, which basically returns a unique value. It works good. I haven't had any duplicates yet, even under heavy load tests.

1 0

kamailio 4.0.4 '408' response
by kamailio＠AaronLux.com 08 Nov '13

08 Nov '13

Below please find a filtered SIP packet capture showing a problem I'm having with callee '408' responses in kamailio 4.0.4. The '408' response occurs after both caller and callee successfully establish 2-way audio. In this setup kamailio is running behind a NAT and both SIP clients connect via WAN and rtpproxy is started with the following command 'rtpproxy -F -l 192.168.1.LAN -A 66.41.221.WAN -s unix:/var/run/rtpproxy.sock -d DBUG:LOG_LOCAL0 -m 10000 -M 20000' Ports 5061 and Ports 10000-20000 are forwarded to the kamailio server and kamailio has TLS enabled. Please let me know if you know if you need more info. Thanks, Aaron # Filtered SIP capture showing only 408 response: http://pastebin.com/pBcZNGA4 # rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Extension 20090810: Support for automatic bridging # kamailio -v version: kamailio 4.0.4 (i386/linux) cabe58 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: cabe58 compiled on 01:07:11 Nov 2 2013 with gcc 4.4.7

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sr-users November 2013