sr-users

sr-users@lists.kamailio.org

8 participants
37523 discussions

(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

3 9

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

2 12

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

2 1

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

2 1

Kamailio out-of-mem
by José Seabra 06 Jul '16

06 Jul '16

I'm running burning tests against kamailio and after some time ~ 2 days I see kamailio out of mem: ERROR: tm [t_lookup.c:1271]: new_t(): ERROR: new_t: out of mem: ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed The result of command kamcmd mod.stats all is: Module: core { sip_msg_shm_clone(494): 10653392 build_req_buf_from_sip_req(2149): 2565480 msg_lump_cloner(978): 2943584 create_avp(175): 969736 xavp_new_value(94): 858469320 counters_prefork_init(207): 16896 cfg_clone_str(130): 88 cfg_shmize(217): 744 init_pt(110): 8 init_pt(105): 8 init_pt(104): 3080 dtrie_insert(157): 240 dtrie_insert(148): 48 dtrie_init(60): 160 dtrie_init(51): 32 register_timer(1011): 360 cfg_register_ctx(47): 32 init_tcp(4635): 8192 init_tcp(4629): 32768 init_tcp(4621): 8 init_tcp(4614): 8 init_tcp(4607): 8 init_tcp(4601): 8 init_tcp(4589): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(437): 16384 init_dst_blacklist(430): 8 timer_alloc(514): 96 init_dns_cache(366): 8 init_dns_cache(358): 16384 init_dns_cache(351): 16 init_dns_cache(345): 8 init_timer(283): 8 init_timer(282): 16384 init_timer(281): 8 init_timer(280): 8 init_timer(269): 8 init_timer(237): 8 init_timer(220): 278544 init_timer(219): 8 init_timer(206): 8 cfg_child_cb_new(830): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 shm_core_lock_init(153): 8 Total: 875992224 } Module: dmq { alloc_job_queue(229): 128 shm_str_dup(723): 24 build_dmq_node(156): 448 add_peer(67): 168 mod_init(243): 8 mod_init(237): 96 init_dmq_node_list(66): 24 init_peer_list(33): 24 Total: 920 } Module: auth_db { Total: 0 } Module: dialog { init_dlg_table(284): 131088 init_dlg_table(276): 8 init_dlg_table(269): 8 init_dlg_table(264): 8 init_dlg_timer(57): 8 init_dlg_timer(48): 32 new_dlg_profile(187): 624 Total: 131776 } Module: rr { Total: 0 } Module: xlog { Total: 0 } Module: kex { pkg_proc_stats_init(79): 1056 Total: 1056 } Module: db_postgres { lock_set_alloc(70): 80 Total: 80 } Module: pv { Total: 0 } Module: textops { Total: 0 } Module: htable { ht_cell_new(183): 1104 ht_init_tables(381): 8388608 ht_add_table(278): 928 Total: 8390640 } Module: sqlops { Total: 0 } Module: tm { build_uac_req(1543): 848 prepare_new_uac(479): 28880 _reply_light(542): 751504 build_local_reparse(341): 1168568 relay_reply(1884): 643088 build_cell(317): 9826624 prepare_new_uac(524): 8112 prepare_new_uac(509): 10192 init_tm_stats_child(60): 5632 insert_tmcb(137): 136 init_tmcb_lists(74): 16 init_tmcb_lists(72): 16 init_hash_table(467): 2097152 Total: 14540768 } Module: usrloc { new_urecord(65): 136 new_ucontact(98): 1096 new_urecord(58): 288 shm_str_dup(723): 1200 build_stat_name(51): 64 new_udomain(93): 40960 new_udomain(86): 48 new_dlist(573): 16 new_dlist(565): 32 register_ulcb(94): 32 init_ulcb_list(45): 16 lock_set_alloc(70): 4112 Total: 48000 } Module: tmx { tmx_check_pretran(250): 864 tmx_check_pretran(271): 1216 tmx_init_pretran_table(90): 256 Total: 2336 } Module: sl { init_sl_stats_child(125): 4224 sl_startup(83): 8 init_sl_stats(110): 8 Total: 4240 } Module: carrierroute { rule_fixup_recursor(584): 8 shm_str_dup(723): 56 add_route_rule(74): 152 add_route_flags(225): 48 create_domain_data(83): 32 create_carrier_data(59): 8 create_carrier_data(50): 64 load_route_data_db(295): 8 load_domain_map(182): 48 load_domain_map(171): 24 load_carrier_map(126): 48 load_carrier_map(115): 24 reload_route_data(172): 64 init_route_data(74): 8 Total: 592 } Module: dispatcher { reindex_dests(600): 1008 add_dest2list(350): 168 add_dest2list(324): 4200 init_data(204): 16 init_data(195): 16 mod_init(309): 8 mod_init(307): 8 ds_ping_active_init(102): 8 Total: 5432 } Module: db_text { dbt_init_cache(81): 256 dbt_init_cache(69): 8 dbt_init_cache(54): 8 Total: 272 } The result of command kamctl shmem is: shmem:fragments = 6353 shmem:free_size = 406208 shmem:max_used_size = 8464105472 shmem:real_used_size = 8463699264 shmem:total_size = 8464105472 shmem:used_size = 5190698208 The server has 8G of shared memory allocated, and the issue happens more or less after 2 days running tests. I'm using sipp to generate traffic, it is sending 500CAPS. There is any possibility of memory leak on Kamailio 4.4.0? Best Regards -- José Seabra

2 11

Kamailio blocked when using with Radius accounting when connection is lost with the database.
by Mx R 06 Jul '16

06 Jul '16

Hi there. I have the following scheme. 1. SIP server is running kamailio with module "acc", radiusclient-ng and freeradius. 2. DB server with MSSQL. The problem is the following, if the connection is lost with the database, no one can make calls. Even after the connection to the database is restored. We have to restart the radius and kamailio. And only after that can make calls. Kamailio logs rc_send_server: no reply from RADIUS server localhost: 1813 ERROR: acc [acc.c: 720]: acc_rad_request (): radius-ing failed Radius logs. Error: rlm_sql_unixodbc: SQL down 08S01 [unixODBC] [FreeTDS] [SQL Server] Write to the server failed Below are links to a similar problem. http://lists.opensips.org/pipermail/users/2012-March/021100.html http://lists.opensips.org/pipermail/users/2012-March/021102.html http://lists.opensips.org/pipermail/users/2012-March/021105.html http://lists.opensips.org/pipermail/users/2012-March/021128.html

1 0

Simple Kamailio configuration for Asterisk load balancing?
by Tickling Contest 06 Jul '16

06 Jul '16

Hello, I am beginning to front my Asterisk cluster with Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for this. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for Kamailio with the following features: (a) Support an arbitrarily large number of Asterisk servers (say, upto 10). (b) Offload SIP registration to a realtime/mysql DB used by the PBXs. Please also let me know if I should have to change my Asterisk PBX config in any way for this to happen. On a very related note, while I appreciate the fact that you need to _really_ understand SIP to configure Kamailio, it should be possible to get this setup by just running a script or GUI. It is difficult to find information on how to load balance multiple PBXs behind Kamailio. Thanks!

3 5

[rtprpoxy LB]
by Noriyuki Hayashi 06 Jul '16

06 Jul '16

Hi, Thank you for be used great KAMAILIO. Regarding rtpproxy LB. I configured as below. modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.136:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.137:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.138:7722") modparam("rtpproxy", "rtpproxy_disable_tout", 20) modparam("rtpproxy", "rtpproxy_tout", 2) modparam("rtpproxy", "rtpproxy_retr", 2) I could not restart kamailio when one rtpproxy server is down. Does anyone tell this solution by the parameter or somthing ? I am looking forward from anyone. Thank you so much. Noriyuki Hayashi

1 0

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by tegamjg 05 Jul '16

05 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)) That is what i was afraid of, so you are not handling kamailio users in the subscribe table of kamailio database but in sipusers table of asterisk database, is that correct? > > 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

1 0

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail
by Daniel Tryba 05 Jul '16

05 Jul '16

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)) 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.

1 0

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