sr-users August 2011

sr-users@lists.kamailio.org

70 participants
108 discussions

Lua script calling rewriteuri
by Sharif Tanvir Rahman 14 Jun '12

14 Jun '12

Hello, We plan to use Kamailio as redirect server. Corresponding contact list is to be fetched by a Lua script from DB. The problem is about calling the required 'rewriteuri' function: 1. From lua script. both 'sr.rewriteuri()' and 'sr.modf("rewriteuri", ...)" fails to write the 'contact' field. 2. I tried an alternative way; pushing the value from lua (e.g. sr.pv.sets("contact", "sip:...")) and got the value in the Kamailio.cfg ($var(contact)). But can't call the rewriteuri function with this variable's value as argument. This seems to be a general problem of calling a function from .cfg with a variable's value as parameter. Any help much appreciated. -- Sharif

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Consulting needed
by Pete Ashdown 14 May '12

14 May '12

I have been trying to accomplish a couple tasks with Kamailio over the past month with no luck. What I need is a bit of one-on-one training with someone who knows the lay of the land. If you do this kind of consulting and can use Skype with possibly a shared-screen terminal, please drop me an email with your rate.

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[Kamailio-Users] Converting rtpproxy recorded .rtp file to .wav file
by Vikram Ragukumar 18 Nov '11

18 Nov '11

Hello, I used Kamailio+rtpproxy to record a session and rtpproxy outputs the following files long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, long_file_name.o.rtcp http://www.rtpproxy.org/wiki/RTPproxy/FAQ From the Rtpproxy FAQ above, i tried to extract the audio using rtpbreak and sox. rtpbreak -W -r long_file_name.a.rtp rtpbreak -W -r long_file_name.o.rtp The above commands generate rtp.0.0.raw, rtp.1.0.raw. Then when i run sox using sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm -s out.wav i get the following errors : sox: invalid option -- - sox: -c must be given a number Is there a switch/anything else that i am missing ? Thanks in advance, Regards, Vikram.

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Re: [SR-Users] OPENSER MIB
by Daniel-Constantin Mierla 17 Oct '11

17 Oct '11

Hello Stefan, On 3/14/11 11:03 AM, Stefan Tiedje wrote: > Thanks for the answer. > Maybe I have some older versions of the OPENSER-MIB and the other > related MIB's since I could not find the counter you pointed at. I'm > using a MIB browser for reading the MIB's. > Is the suggested counter "expired dialogs" added in a specific release > of Kamailio? Which? We use Kamailio 3.0.2. I used Kamailio and recommend using it sine it has the latest commits for stability. However, what I wrote before is pretty much not related to the version. There is a counter that tracks the processed dialogs, but seems it is not exported by default through snmpstats module. The statistics counter is named "processed_dialogs", implemented by dialog module. You can dump all internal statistics through kamctl or via xmlrpc command, but probably to export it through snmpstats you may need to extend the mibs and the code of the module. I just grepped the sources of snmpstats module to see what dialog statistics it is exporting: $ grep -n _dialogs modules_k/snmpstats/* | grep get_statistic modules_k/snmpstats/alarm_checks.c:83: num_dialogs = get_statistic("active_dialogs"); modules_k/snmpstats/snmpObjects.c:404: int result = get_statistic("active_dialogs"); modules_k/snmpstats/snmpObjects.c:424: get_statistic("active_dialogs") - modules_k/snmpstats/snmpObjects.c:425: get_statistic("early_dialogs"); modules_k/snmpstats/snmpObjects.c:443: int result = get_statistic("early_dialogs"); modules_k/snmpstats/snmpObjects.c:459: int result = get_statistic("failed_dialogs"); modules_k/snmpstats/snmpObjects.c:508: int num_dialogs = get_statistic("active_dialogs"); Perhaps when the snmpstats was developed the dialog module didn't export the statistics counter of "processed_dialogs" and then it was not updated. Now, what I tried to say is that if the "processed_dialogs" counter is not available through snmpstats (and it is not now after grepping the sources) you can get its value from another application through "kamctl get_statistics all" or XMLRPC command for all of the existing kamailio releases. Upcoming one we will look to implement the export through snmpstats as well. If you have time to do it and send us a patch, we will gladly commit it to source tree in our GIT repository. Cheers, Daniel > Do you have the MIB name for the "expired dialogs" counter. I will > look for that in my version of OPENSER MIBS. > Important, do you have a link to where MIB files can be downloaded for > Kamailio 3.0.2? > Below follows an excerp from one of the MIB's. Is it old, I don't know? > > -- *********************************************************************** > > -- OPENSER-MIB: OPENSER MIB > > -- > > -- Date of Creation: Januay 2006 > > -- > > -- This MIB provides information related to the OpenSER SIP Router. > > -- > > -- Copyright (c) The Internet Society (2006) > > -- Ammendments (c) Soma Networks, Inc. (2006) > > -- > > -- All rights reserved. > > -- ***************************************************************** > > /Stefan > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* den 14 mars 2011 10:16 > *To:* Stefan Tiedje > *Cc:* sr-users(a)lists.sip-router.org > *Subject:* Re: [SR-Users] OPENSER MIB > > Hello, > > On 3/14/11 9:42 AM, Stefan Tiedje wrote: >> Hi, >> In the Kamailio OPENSER-MIB there is the counter >> "openserTotalNumFailedDialogSetups". This is a Counter32. >> The description is: >> "The total number of calls that failed with an error. The >> following codes define a failed call:" >> *Question:* >> >> * I'm looking for the corresponding counter to >> "openserTotalNumFailedDialogSetups" who counts successful >> Dialog setups of Counter32 type. Does it exist? >> * If not, does it exist a work around? >> * Where in the code can the new suggested counter be added? >> * Something else???? >> > > the dialog module counts the number of processed dialogs, see: > http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2966360 > > There is no counter currently inside dialog module exporting exactly > the number of successfully setup dialogs, it should not be hard to do > it, though. Using the above and the number of failed and expired > dialogs, you can actually get the number of successful dialogs. > > Dialog module being the one that tracks SIP dialogs, therefore being > able to count them, now I don't know if snmpstats module exports all > the counters from dialog module. I setup snmpstats just few weeks ago > and works perfect on Ubuntu/Debian servers, but I had no need to check > dialog module counters. > > Note that you can get the list of all internal statistics via kamctl: > - kamctl fifo get_statistics all > > Or via XMLRPC if you need them remotely in another application. > > Another option is to define your statistics with statistics module. > Knowing that in SIP a successful call dialog means 200ok reply to an > INVITE transaction, you can count it in the onreply_route[abc] that > you arm for relayed transactions with t_on_reply("abc"). > > Hope these help you, > Daniel > >> Suggestion for the new counter is a name like: >> "openserTotalNumSucceededDialogSetups". It has a counter32. >> Description: "The total number of calls that succeeded" >> I know that there are the counters openserCurNumDialogs, >> openserCurNumDialogsInProgress and openserCurNumDialogsInSetup but >> these are of Gauge type who only reflects the current situation. >> These Gauge counters can't be used together with a Counter32 counter. >> That don't mix. The calculation done for the counter >> "openserCurNumDialogsInProgress" should be used where every new >> dialog setup is added to the new suggested counter. A counter of 32 >> should cover a great deal of connections. These counters are usually >> read, if used, every 15 minutes or 1 hour. >> *Rationale:* >> The reason for the new counter is that a calculation between >> succeeded and failed dialog setups can be done and be used for SLA >> agreements. Without this, its hard to make any customer versus >> provider agreements. >> /Stefan >> PS. Ask if anything is unclear and I need an answer rapidly. >> >> >> _______________________________________________ >> 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 > > -- > Daniel-Constantin Mierla > http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

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Multiple INVITEs and discarded call_control
by Mino Haluz 07 Oct '11

07 Oct '11

Hi, my kamailio server is receiving from some customers 3 identical INVITEs when call is initiated (separated by 200ms). Those 3 INVITEs are making a big problem with call_control: WARNING: call_control [call_control.c:1156]: dialog to trace controlled call was not created. discarding callcontrol. That is why, the prepaid limit is not working at all in this case. This way the user can hack the prepaid protection of the account. Otherwise the call_control is fuilly functional. Anybody experienced the similar problem? If so, how to resolve it? Thanks, Mino Haluz

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Balancing incoming traffic (DID)
by Anto 05 Oct '11

05 Oct '11

Hello It presents a problem because I do not know how to assimilate the implementation. Until now we had a design of a proxy to balance out between several gateway, but it will add another proxy for another service (other users) and use the same gateway output or perhaps different (although not that be better if divide or unify the gateway). The problem as I have with the routing entry, which is not very well how to implement it. Now we have the design like this: -GW ----\ / - CARRIER1 USER ------ KAMAILIO ----- GW ---- + \ - CARRIER2 -GW ----/ And becomes like this: -GW ----\ / - CARRIER1 USER ------ KAMAILIOSERVICE1 ----- GW ---- + \ - CARRIER2 -GW ----/ -GW ----\ / - CARRIER1 USER ------ KAMAILIOSERVICE2 ----- GW ---- + \ - CARRIER2 -GW ----/ Or maybe using the same gateway. The problem is that each carrier provides DID number (I use the same for both services). Kamailio had thought to put a gateway between the carriers and that they did check the service and send it to the corresponding gateway (in the case of the gateway separately), or put it in the proxy gateway and making checking for routing to proxy (in the case of using the same gateway). What would be better? Any recommendations on how to put this into practice?. The biggest question that I believe is the routing table, as if this new proxy routes the traffic into and leaves her in the field via ip, also pass through the output data (I have doubts whether SIP level, following standard, you can not pass through the exit) and want it to be direct (USER -> Proxy -> GW -> CARRIER), just past the entrance. Anyone could advise me where I could find more information to set up this scenario?. thanks and sorry for bothering you with this question ;-). best regards

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Open source HLR (Home Location Register).
by Henrik Aagaard Sørensen 16 Sep '11

16 Sep '11

I know this is the mailing list for Kamailio, but hopefully someone can help me out as it is "kinda" related to VoIP etc. Does anyone know any open source / free HLR implementation?

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parse errors in rr-callback of path module
by Andreas Granig 14 Sep '11

14 Sep '11

Hi, path module calls register_rrcb when "use_received" is set to 1 to extract the "received=" param and set its content as dst-uri if present. Now I've noticed that when e.g. a message has a pre-loaded route like "Route: <sip:mydomain.com;lr>", then kamailio (at least 3.1.3) spits the following error message: /usr/sbin/kamailio[23953]: ERROR: <core> [parser/parse_param.c:481]: parse_params(): Invalid character, ; expected /usr/sbin/kamailio[23953]: ERROR: path [path.c:213]: failed to parse route parametes I was trying to reproduce this but without much success, it sometimes happens and sometimes not, but I can't yet put the finger on the reason. Any ideas? Andreas

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no publish from pua/pua_mi to outbound proxy
by Juha Heinanen 12 Sep '11

12 Sep '11

i started to test sr presence and for some reason sr does not send out any publish request when i issue one via pua_mi. i have in route [xmlrpc_requests]: xlog("L_INFO", "Handling XMLRPC $rm from <$si> with body <$(rb{re.subst,/[\n\r\t]//g})>\n"); set_reply_no_connect(); set_reply_close(); dispatch_rpc(); syslog shows that sr receives xmlrpc request: Aug 29 19:42:13 sip /usr/sbin/pres-serv[2118]: INFO: Handling XMLRPC POST from <127.0.0.1> with body <<?xml version='1.0'?><methodCall><methodName>mi</methodName><params><param><value><string>pua_publish</string></value></param><param><value><string>sip:jh@vm.test.fi</string></value></param><param><value><int>7776000</int></value></param><param><value><string>message-summary</string></value></param><param><value><string>application/simple-message-summary</string></value></param><param><value><string>.</string></value></param><param><value><string>a.1313821931.6178.2.0</string></value></param><param><value><string>P-Flags: 0</string></value></param><param><value><string>Messages-Waiting:yesVoice-Message: 1/1 (0/0)</string></value></param></params></methodCall>> and wireshark shows that sr xmlrpc server responds to it with 200 ok. but after than nothing happens, i.e., according to wireshark no publish request is sent out. i have pua and pua_mi modules loaded and outbound proxy defined: modparam("pua", "outbound_proxy", "sip:127.0.0.1:5082") there is no error messages in syslog. any idea why sr does not send publish to outbound proxy? this used to work with opensips. -- juha

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Siremis User Role permissions
by Camila Troncoso 07 Sep '11

07 Sep '11

Hi, I have a question about the assignment of Role permissions to limit some Users privileges in Siremis. I defined a user , and then a Role denying some actions, but I couldn’t deny the action from de SER menu . The Role permissions only show the modules on Application Menu from Administration Tab . What if I want to deny access to some items inside SER Menu , like modifying the ACL services or /and Rounting services? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo VoissNet S.A. *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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sr-users August 2011