sr-users July 2016

sr-users@lists.kamailio.org

82 participants
126 discussions

A parameter with a life span of the entire dialog
by Jay Li 07 Jul '16

07 Jul '16

Dear All, I'm trying to set a few parameters for an "INVITE", and access their values in the failure routine if the call fails. $var() doesn't seem suitable for my purpose as it only lasts for one process. I think if I set a $var() in "INVITE", it will be gone when I receive the failure response. $sht() seems able to last long but it will be accessed by all processes which might include ones not belonging to the same dialog. I wonder if somebody would give me some suggestions here to handle such situations. Does Kamailio provide a mechanism to set up a parameter with a life span of the entire dialog? Thanks. Regards,Jay

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call_control not found error
by Infinicalls Infinicalls 07 Jul '16

07 Jul '16

Hi, I am trying to implement a prepaid system and get this error " ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <call_control> found " while adding modparam("call_control", "disable", 1). I have already included pv and dialog in the modules list. Any idea? regards Ganesh Kumar

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t_check_trans() exits route execution if transaction is found
by Sebastian Damm 07 Jul '16

07 Jul '16

Hi, on one of our systems we are checking a few times whether a transaction already exists. We noticed that route execution stops if the transaction is found. And after looking into the code, we found that it seems to be intended like this. >From tm.c: int t_check_trans(struct sip_msg* msg) { if (msg->first_line.type==SIP_REPLY) { [..] } else { switch(t_check_msg(msg, 0)){ [..] case 1: /* found */ [..] return 0; /* exit from the script */ } } } Can anybody explain, why script execution should stop when a running transaction is found? I would expect that then the function returns 1 so I can use it in my if statement and continue with my routing script. Thanks already for an explanation, Sebastian

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Kamailio compact headers ?!
by Anonim Stefan 07 Jul '16

07 Jul '16

Hi, Is there a parameter to be enabled such that Kamailio will append headers in compacted form? (I'm thinking of Via: and Record-Route:) Thanks, Stefan

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Kamailio and Mongodb
by C. 06 Jul '16

06 Jul '16

Hi, I am still trying to get Kamailio to work with MongoDB. Followed instructions here https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb <https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb>, but I cannot get things to install. Are there any instructions on the exact configuration changes that need to be made to support Mongodb? For example, kamctl points to MySQL, etc Thanks! Chuks.

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Locally originated BYE dialog database handle
by Dirk Teurlings - Signet B.V. 06 Jul '16

06 Jul '16

Hi, There's something going on with a locally originated BYE and the dialog module running with a database backend. Here's the scenario. Call gets setup, and is being monitored by outside processes (non-Kamailio). After a while these processes determine the call is taking too long (or too expensive), so it has to be killed. To do this we're using the kamcmd dlg.end_dlg with h_entry and h_id as parameter. This works to end the call, but is throws some weird issues into the routing. First off, the event_route[dialog:end] is not thrown 99% of the time, though I would expect it to. Second, This error pops up in the log /usr/sbin/kamailio[25094]: ERROR: dialog [dlg_db_handler.c:199]: use_dialog_table(): invalid database handle As if the command issued by kamcmd isn't able to get connected to the dialog database. In any case, this error results in the dialog entry for this call not being removed from the database. So this piles up after a while. Anybody any thoughts on this? Cheers, Dirk

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(no subject)
by John Petrini 06 Jul '16

06 Jul '16

Hello, I'm trying to append a branch, assign it a q value so it has priority and serialize the new branch and the main branch. The goal is to send the new branch off to our CNAM provider, capture the CNAM in their reply to an $avp then drop the branch, stick the cnam in the main branch and allow the call to continue as normal. We do this currently using OpenSIPS with serialize_branches(1) but the of load_contacts() behavior seems to be different than serialize_branches. route[CNAM_DIPS] t_on_branch("CNAM_DIPS"); t_on_reply("CNAM_DIPS"); t_on_failure("CNAM_DIPS"); t_load_contacts(); t_next_contacts(); append_branch("sip:2223334444@22.33.444.55:5600;trans-type=5", "0.5"); relay(); break; Two problems. First when t_next_contacts is called I get the following error: ERROR: pv [pv_branch.c:58]: pv_get_branchx(): error accessing branch [0] So it looks like the branches aren't actually being loaded? Second problem is that despite the first error my call goes out on the branch I appended and I get a reply from the CNAM provider but Kamailio forwards that reply back to the phone causing the phone to send a second invite. In OpenSIPs we just called drop, to drop the branch and let the main branch continue but this doesn't seem to work in Kamailio. I've read and reread the documentaion but it doesn't seem to behave the way the documentation says it should. Is there something I'm missing? Is there a better way to accomplish this in Kamailio? Thank You, ___ John Petrini

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Issue with userblacklist module after 4.4.2 upgrade
by Tim Balmer 06 Jul '16

06 Jul '16

Hello, We are running Kamailio version 4.4 and have our sipproxy servers configured to do auto updates within te same major version based on a 4 weekly schedule (we have 4 sipproxy / Kamailio servers). So this morning as expected the first of our sipproxy servers automatically upgraded from 4.4.1 to 4.4.2, unfortunately after this update we have issues with the userblacklist module. The problem is that with the 4.4.2 release every number is now matching our userblacklists although it is not really blacklisted. After some digging in the logs I found this is probably because the userblacklist module is looking for the wrong database table. With every check_user_blacklist call I see errors like this appearing in the logs: Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Table 'kamailio.<number>' doesn't exist (1146) Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: userblacklist [db.c:68]: db_build_userbl_tree(): error while executing query. Jul 5 13:24:03 tel-dev-01 [err] /usr/sbin/kamailio[2834]: ERROR: userblacklist [userblacklist.c:277]: check_user_list(): cannot build d-tree I replaced the real number with <number> in the log output but it seems like the userblacklist module is now looking for a database table with the name of the number to check against. On our development server I tested setting the global userblacklist_table parameter and also passing the table parameter within the check_user_blacklist call but both are not making any difference. After downgrading the sipproxy back to 4.4.1 everything is working as expected again so it looks to me like a bug in the 4.4.2 release. Is anyone else experiencing the same issues after the 4.4.2 upgrade or is it just me doing something wrong? Best Regards, Tim Balmer

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Kamailio summer-of-documentation - join the effort
by Olle E. Johansson 06 Jul '16

06 Jul '16

Hi! Looking for a summer-of-documentation project? Which README is the best? Which one is so full of Swedish that you totally fail to parse it? Check our documentation pages, suggest improvements on the mailing lists (or here on Facebook) and give us feedback! This is something you can spend your lazy summer days on - a nice cool drink, cool sunshades and Kamailio documentation! We constantly work to update our README files for every module and the cookbooks for the Kamailio core, the transformations and pseudovariables. But we’re developers, not casual administrators and propably fail to see some obvious things missing. Please tell us (feel free to use the mailing lists). And if you can - but not required at all - you can suggest rewrites in the XML source and upload to github. Let’s keep our docs up to date and make sure they really assist our users! http://kamailio.org/docs/modules/devel/ Have a great summer! Regards, /O

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Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by tegamjg 06 Jul '16

06 Jul '16

El 05/07/2016 11:36, Daniel Tryba <d.tryba(a)pocos.nl> escribió: > > Please keep the mailinglist in the loop, so everybody might benefit from > our ramblings :) > > > Still there are few things i dont understand, i am not using asterisk > > just as a voicemail server since they are actually handling also the > > calls passing first from kamailio and being load balanced to those > > asterisk boxes. May i still use call forwarding as you are using it? > > (Both asterisk have a shared storage with a clustered filesystem, so > > both will be able to see voice messages) > > Yes I think so. I use a seperate machine for voicemail but I see no > problem with other uses (I used to use it for playback of messages and > transcoding ebtween incompatible endpoints). > > By using the prefixes in kamailio to the username in $ru I have in the > extensions.conf: > > exten => _tovm-.,1,NoOp(leave voicemail) > exten => _tovm-.,n,Answer() > exten => _tovm-.,n,Set(CHANNEL(language)=nl) > exten => _tovm-.,n,Voicemail(${EXTEN:5},us) > exten => _tovm-.,n,Playback(Goodbye) > exten => _tovm-.,n,Hangup() > > exten => _getvm-.,1,NoOp(read voicemail) > exten => _getvm-.,n,Set(CHANNEL(language)=nl) > exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) > exten => _getvm-.,n,Hangup() > > > The other question is that i actually though that you need asterisk to > > have users configured in sipusers realtime table to associate their > > mailboxes, which i dont have since those users are stored in the > > subscriber table of kamailio. So am i still able to configure > > voicemail like you are doing it by syncing with the voicemail table?, > > i really hope so haha > > I forgot that fact. So yes I have a realtime sip users list (with > host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio > username, no password (this machine is not directly accessible from > outside)) Sorry, I think i rushed the last answer but if you could answer that one would be nice How are you handling the calls? Just with kamailio/rtpproxy? Because i am also using asterisk for calls with dial application and for nat issues (with kamailio behind nat) i am using also kamailio/rtpproxy for outside. All this with just handling users (registration and location) in the subscribe and location table of kamailio. That is why i am not using sipusers table of asterisk because of nat was behaving weird using it that way. Could it be possible to use both tables without expecting a different behaviour? Or is not, in the end, a good idea and i need to keep users in sipusers table? > > You might not be able to have endpoints able to subscribe to > notifications due to this. I baked something inspired by: > http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… > that appears to work for me. > > _______________________________________________ > 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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sr-users July 2016