- sr-users - kamailio.org

CNXCC compilation errors
by Infinicalls Infinicalls 07 Jul '16

07 Jul '16

Hi, Thanks. Well I tried it before also but could not proceed with since I got the compilation errors. Now also I am struck up with the same error mentioned in http://lists.sip-router.org/pipermail/sr-users/2016-January/091574.html Has anybody found a solution to this? Mine is Ubuntu 12.x LTS regards Ganesh Kumar On 7/7/16, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > look at cnxcc module for implementing a prepaid system. There should be > also a presentation about it at one of the kamailio world conferences -- > see kamailio.org/events/ > > Cheers, > Daniel > > > On 07/07/16 11:33, Infinicalls Infinicalls wrote: >> Hi Daniel, >> >> Thanks. I figured out later :-) >> >> I am actually trying to implement prepaid system with dialog and call >> control. In Siremis, after a call the duration and other details get >> updated but not the cost. This is inspite of having rates and >> kamailio_cdr / kamailio_rating() in cron. >> >> Any idea? >> >> regards >> Ganesh Kumar >> >> On 7/7/16, Daniel Tryba <d.tryba(a)pocos.nl> wrote: >>> On Wed, Jul 06, 2016 at 11:01:17PM +0530, Infinicalls Infinicalls wrote: >>>> " 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? >>> Try loading the module first :) >>> >>> https://www.kamailio.org/wiki/cookbooks/4.4.x/core#loadmodule >>> >>> >>> _______________________________________________ >>> 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 > > -- > Daniel-Constantin Mierla > http://www.asipto.com - http://www.kamailio.org > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > > _______________________________________________ > 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 > -- -- http://www.infinicalls.com

1 0

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

2 1

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

3 3

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

2 1

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

6 10

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.

1 0

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

2 2

(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

dispatcher module not fetching attrs col from DB
by Jayesh Nambiar 05 Jul '16

05 Jul '16

Hi, I'm trying to use the dispatcher module with algo 9 for weighted distribution. I have a column attrs in the dispatcher table with value in the format "weight=90" for my destination addresses. But somehow the ds_list command always shows attrs as blank. I checked that kamailio doesn't fetch the attrs column at all; neither while starting up nor while calling ds_reload. This is the query that kamailio sends to my DB while starting up or calling the fifo ds_reload: select `setid`,`destination`,`flags`,`priority` from `dispatcher` I dont even see an attr column modparam in the documentation although kamailio didnt throw an error when I defined attr_col modparam for dispatcher module. Is there a different way to tell kamailio to fetch the attrs from the DB?? I'm using the 4.4.1 version. Thanks for any suggestion or pointers to solve this. - Jayesh

2 4

Kamailio v4.2.8 Released
by Daniel-Constantin Mierla 05 Jul '16

05 Jul '16

Hello, Kamailio SIP Server v4.2.8 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.7. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.8 (or to 4.3.x/4.4.x series). Important: this version marks the end of planned releases from branch 4.2. From now on the focus is on maintaining the branches 4.4 and 4.3 for stable releases. For more details about version 4.2.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/07/kamailio-v4-2-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.4, at this moment with its latest release v4.4.2. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

[SR-USERS] Kamailio issues with Asterisk Voicemail
by TEG AMJG 05 Jul '16

05 Jul '16

Dear list: I am quite new to Kamailio and i am really struggling to get some answer to some questions i have been trying to get answers by reading documentation and other questions or solutions around internet. Even though i am integrating Kamailio with two Asterisk boxes with the module dispatcher, i didnt follow the quite popular guide of Asterisk Kamailio Realtime Integration ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). The reason why i didnt is that it actually behaved weird since i wanted Kamailio to be a stateful proxy and also wanted Kamailio to act as a Registrar Server and Location Server using subscriber and location tables of kamailio database, keeping both Asterisk strictly as Media Servers. Also i have been able to solve some NAT Traversal issues using RTPProxy and not using those Asterisk boxes with their nat or qualify parameters to solve NAT issues. Most of my solution started following this guide ( http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html) which let me handle registration and location with Kamailio and also changed it putting kamailio behind NAT. Now to the real question, is there anyway to include Asterisk voicemail functionality with the solution I already mentioned? the question is because, as I far as I know, Asterisk needs to associate mailboxes to its users. So i believe that i have to practically change most of my solution and find a way to make Kamailio register and locate users with the Realtime integration of Asterisk and map the registration accordingly to the table fields. But it destroys most of my NAT and register solution since i also configured one of the private IPs as a peer in Asterisk sort of like in that guide [Kamailio] type=friend host=192.168.1.244 port=5060 disallow=all allow=gsm allow=g729 allow=alaw allow=ulaw Sorry for making the question so long and to be quite ignorant with some of the concepts involving Kamailio with Asterisk Thanks Alejandro

2 1

Increase diversion counter
by Igor Potjevlesch 05 Jul '16

05 Jul '16

Hello, I'm looking for the best way to increase the Diversion counter. If my Kamailio is the first to generate a call forward I can manage this with static value into suffix params. But, if the Diversion counter is already set, I don't see how to handle this properly. Someone has already work on this? Regards, Igor.

3 5

msgpack/format - new module?!
by Anonim Stefan 05 Jul '16

05 Jul '16

Hi, I'm thinking of creating a new module called "msgpack" or "format" which will export config msgpack_serialize() and msgpack_deserialize() for a given input. Is a new module the way to go, or do you think of a better place for this? Thanks, Stefan

2 3

Releasing Kamailio v4.2.8
by Daniel-Constantin Mierla 05 Jul '16

05 Jul '16

Hello, I am planning to release tomorrow the v4.2.8 out of branch 4.2 to mark the end of official packaging for 4.2.x series. From now on, the focus will be 4.4.x and 4.3.x as stable branches. I just backported what I found relevant from branch 4.3 to 4.2 -- if anyone finds some other commits that need to be backported, reply to mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

register_dlgcb usage example
by Alexandr Dombrovsky 05 Jul '16

05 Jul '16

Hi! Where I can find any example of usage for this function: *register_dlgcb ?*

2 1

Modify VIA header in replied message
by Huynh Quoc Bao 05 Jul '16

05 Jul '16

Hi all, As title, i want to add some additional parameters to the topmost via header in replied message. example. REGISTER ==> Kamailio ===> reply 200 OK. So in 200 OK message, i will add some additional parameters. And the same for the other message: INVITE, SUBSCRIBE, NOTIFY, MESSAGE message. -- Brs/Bao

2 3

INFO: <core> [main.c:809]: sig_usr(): signal 15 received
by Aqs Younas 04 Jul '16

04 Jul '16

Greetings Again List. After being unable to use ndb_cassandra with kamailio I tried to use app_python. Below is relevant configuration. version: kamailio 4.4.2 (x86_64/linux) 892ad6 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 892ad6 compiled on 07:50:03 Jul 1 2016 with gcc 4.9.2 modparam("app_python", "script_name", "/usr/local/kamailio/etc/kamailio/handler.py") request_route { ..... python_exec("handler"); ..... } And i am using example provided in app_python module. *But i soon as i import Cassandra driver in script*. Kamailio terminate with there logs. Jul 1 19:57:19 debian ./kamailio[8643]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(20) Jul 1 19:57:19 debian ./kamailio[8642]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(19) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: tmx Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: tmx [tmx_mod.c:275]: child_init(): rank is (-4) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: sl Jul 1 19:57:19 debian ./kamailio[8616]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Jul 1 19:57:19 debian ./kamailio[8642]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa78988 and 0xabc988 Jul 1 19:57:19 debian ./kamailio[8642]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa787c0, 40, 1, (nil)), fd_no=0 Jul 1 19:57:19 debian ./kamailio[8644]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 21: kex Jul 1 19:57:19 debian ./kamailio[8645]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (22) Jul 1 19:57:19 debian ./kamailio[8640]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (17) Jul 1 19:57:19 debian ./kamailio[8639]: DEBUG: mi_rpc [mi_rpc_mod.c:104]: child_init(): initializing child[-2] for rpc handling Jul 1 19:57:19 debian ./kamailio[8646]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 23: tm Jul 1 19:57:19 debian ./kamailio[8641]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 18: corex Jul 1 19:57:19 debian ./kamailio[8635]: INFO: app_python [mod_Logger.c:170]: logger_LM_INFO(): test.child_init(-1) Jul 1 19:57:19 debian ./kamailio[8648]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank -4: usrloc Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD Jul 1 19:57:19 debian ./kamailio[8636]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8621]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8620]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8625]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8623]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8638]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8624]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8632]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8647]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8633]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8637]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8619]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8622]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8626]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8627]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8628]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8629]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8642]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8630]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8635]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8631]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8634]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done Jul 1 19:57:19 debian ./kamailio[8616]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc Jul 1 19:57:19 debian ./kamailio[8616]: DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc I have tried using two different drivers for Cassandra in python but same results. I wonder if i import mysql driver kamailio starts perfectly. Below it my script. #!/usr/bin/python import sys import pymysql *import pycassa* from Router.Logger import LM_ERR from Router.Logger import LM_INFO class test: def __init__(self): LM_INFO('test.__init__\n') def child_init(self, y): LM_INFO('test.child_init(%d)\n' % y) return 0 def handler(self, msg): LM_INFO('test.handler(%s)\n' % (msg.Type)) if msg.Type == 'SIP_REQUEST': if msg.Method == 'INVITE': LM_INFO("Re-writing RURI") msg.rewrite_ruri('sip:0022@192.168.2.24:5073') LM_INFO('SIP request, method = %s, RURI = %s, From = %s\n' % (msg.Method, msg.RURI, msg.getHeader('from'))) LM_INFO('received from %s:%d\n' % msg.src_address) return 1 def mod_init(): return test() Any help is much appreciated. Best Regards, Aqs Younas

1 0

Error while reading from Cassandra Cluster
by Aqs Younas 04 Jul '16

04 Jul '16

Greetings Lists. I am having hard time trying to use ndb_cassandra with kamailio. I have successfully complied the module and kamailio runs without any error. Below is relevant snippet. loadmodule "ndb_cassandra.so" modparam("ndb_cassandra", "host", "192.168.0.103") modparam("ndb_cassandra", "port", 9160) $var(keyspace) = 'kamailio'; $var(column_family) = 'numbers_directory'; $var(key) = 15; $var(column) = 'number'; xlog("L_INFO", "Valus to fetch are $var(keyspace), $var(column_family), $var(key), $var(column) \n"); * if (cass_retrieve("$var(keyspace)", "$var(column_family)", "$var(key)", "prefix", "$var(val_read)") > 0) {* xlog("L_INFO", " ===== ndb_cassandra. Sucess while reading from Cassandra. val_read: \"$var(val_read)\"\n"); } else { xlog("L_INFO", " ===== ndb_cassandra. Error while reading from Cassandra \n"); } Cassandra Table Structure. cqlsh:kamailio> describe numbers_directory; CREATE TABLE kamailio.numbers_directory ( id int PRIMARY KEY, description text, number text ) WITH bloom_filter_fp_chance = 0.01 AND caching = '{"keys":"ALL", "rows_per_partition":"NONE"}' AND comment = '' AND compaction = {'class': 'org.apache.cassandra.db.compaction.SizeTieredCompactionStrategy'} AND compression = {'sstable_compression': 'org.apache.cassandra.io.compress.LZ4Compressor'} AND dclocal_read_repair_chance = 0.1 AND default_time_to_live = 0 AND gc_grace_seconds = 864000 AND max_index_interval = 2048 AND memtable_flush_period_in_ms = 0 AND min_index_interval = 128 AND read_repair_chance = 0.0 AND speculative_retry = '99.0PERCENTILE'; cqlsh:kamailio> select * from numbers_directory; id | description | number ----+-------------+----------- 15 | null | 212456789 When i make a call i get this in logs. Jul 1 19:10:42 debian ./kamailio[8142]: INFO: <script>: Valus to fetch are kamailio, numbers_directory, 15, number Jul 1 19:10:42 debian ./kamailio[8142]: INFO: <script>: ===== ndb_cassandra. *Error while reading from Cassandra* I am stuck at this. Does anybody able to use kamailio with cassandra or can share their configuration? Any pointer on this is much appreciated. Best Regards, Aqs Younas

1 0

Kamailio and Mongodb
by C. 04 Jul '16

04 Jul '16

Hi, I have followed instructions on setting up Kamailio with MongoDB backend, and successfully compiled and installed kamailio with the db_mongo module. It can start successfully. However, when I tried to add users using the command: kamctl add test testpasswd test(a)mysipserver.com <mailto:test@mysipserver.com> It fails because it is trying to default to MYSQL. Here is the result of that command: WARNING: database engine not found - tried 'MYSQL' ERROR: This command requires a database engine - none was loaded When I take a look at the kamctl script, it lists all the other kinds of database engines, except MongoDb, so this is probably why it is failing. How can I use kamctl to create the required Mongodb collections (like users, subscribers, etc) that Kamaiio will then use?My kamailio.cfg seem to be working ok, as I seem to be able to connect to the database (I modified the file to use db_mongo module), and I create the initial collections as listed here - https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb <https://www.kamailio.org/wiki/tutorials/kamailio-and-mongodb> Thanks, Chuks.

2 1

Regarding config-engines
by Alexander Rudenko 04 Jul '16

04 Jul '16

Hi, Question about future config-engines. Will it be possible to use connectors/drivers via the engine script and is it a good Idea? E.g. I have specific database and python driver for that database. May I use simple 'import dbdriver', create connection to the db, use it inside python engine script, etc? In general, in case there is no kamailio connector module for scpecific database, I would use python connector. Thanks in advance! -- *Alexander Rudenko*

2 1

re-Invite in the failure route
by Jay Li 04 Jul '16

04 Jul '16

Dear All, Hopefully I'll be able to get some help here on re-INVITE in the failure route. For example I have a couple of PSTN gateway options. I'd like to try them by priorities. In case the one with top priority couldn't go through(e.g. returned 5XX on INVITE), I'd like to re-INVITE through another provider in the failure route. Not sure what I should do for such cases. Thanks. Regards,Jay

2 3

Issue loading ims_dialog module
by Vikram Chhibber 01 Jul '16

01 Jul '16

Hi All, I am trying to bring-up PCSCF on releae 4.4 and I am getting errors related to loading of ims_dialog module. The mpath is set correctly in kamailio.cfg and ims_dialog.so is present in that path. Interestingly, it is able to load rest of the modules from the same locaiton. ls -lrt /usr/local/lib64/kamailio/modules/ims_dialog.so -rwxr-xr-x 1 root root 953760 Jun 29 16:37 /usr/local/lib64/kamailio/modules/ims_dialog.so Any pointers will be appreciated. -- Unit kamailio.service has begun starting up. Jun 29 17:08:17 kamailio1 kamctl[2914]: ** INFO: Starting Kamailio : Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [sr_module.c:582]: load_module(): could not open module </usr/local/lib64/kamailio/modules/ims_dialog.so Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 169, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <ims_dialog> found Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 313, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <ims_dialog> found Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 314, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <ims_dialog> found Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 315, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <ims_dialog> found Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 316, colum Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [cfg.y:3228]: yyparse(): cfg. parser: failed to find command set_dlg_profile (params 1) Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 1009, colu Jun 29 17:08:17 kamailio1 kamailio[2939]: ERROR: <core> [cfg.y:3228]: yyparse(): cfg. parser: failed to find command set_dlg_profile (params 1) Jun 29 17:08:17 kamailio1 kamailio[2939]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 1077, colu Jun 29 17:08:17 kamailio1 kamailio[2939]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 29 17:08:20 kamailio1 kamctl[2914]: ** ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed Jun 29 17:08:20 kamailio1 systemd[1]: kamailio.service: Control process exited, code=exited status=1 Jun 29 17:08:20 kamailio1 systemd[1]: Failed to start Kamailio SIP Server. -- Subject: Unit kamailio.service has failed Thanks

2 4

Distributed Authentication
by Colin Morelli 01 Jul '16

01 Jul '16

Hey all, I'm running a cluster of Kamailio instances as a proxy/registrar for another cluster of Freeswitch instances. I'm using http_async_client to make HTTP queries to my API to fetch credentials on auth challenges. Kamailio performs generating the header, and validating the result based on the data provided from my API. I'm fairly sure the answer is no, but I was wondering if Kamailio has any mechanism for getting access to the nonce/nc values in the challenges and responses so I can store them somewhere accessible to the whole cluster. Because my instances are transaction stateful, the request that is challenged and the subsequent request with the response may be routed to different instances and I want to validate the nonce correctly. I can move all of this into the API (the digest auth and verification), but my next question would be whether or not there are any APIs for getting access to this information in a structured format, or if I should just shove the whole digest auth header in the request to my API and parse/verify there. Thanks in advance. Best, Colin

2 1

Serialize Branches Based on q value
by John Petrini 01 Jul '16

01 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 a $var 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 and the serialize_branches(1) but the of load_contacts() 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] Second problem is that 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

1 0

Dialplan module doubt
by José Seabra 01 Jul '16

01 Jul '16

Hello there, I'm evaluating if I can use dial-plan and dispatcher module to build a kamailio routing server to route calls based on number dialed(sometimes the number dialed can be Alphanumeric). I'm expecting have lot of dial-plan entries, some of them with manipulations, others without. Anyone can see here any limitation on this implementation? One thing that I have noticed is that dial-plan module enforces me to fill the fields substr_exp and repl_exp even if i don't have any manipulations to apply. I'm also concerning about performance issues with lot of dialplan entries. What are your thoughts about this solution? Thank you -- José Seabra

2 6

ACK / BYE transaction problem
by Oliver Roth 01 Jul '16

01 Jul '16

Hi all Follow scenario Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> gateway kamailio [gw] --> carrier [carr] I get Invites from [c5] with Request ,To, from, contact, pid in national format 0794445566 [lbl] dispatches this to [gw] For the [carr] I need international format. So doing these transactions in [gw] And sending to [carr] in international format Request, to, from, contact, ... => 417794445566 Everything ok Then I get a 100, 183 and even 200 from [carr] Ack is coming from [c5] to [lbl] and [gw] - but then it stocks The ACK is not sent to the [carr] I kamailio log I see DEBUG: RFC3261 transaction matching failed DEBUG: t_lookup_request: no transaction found So for me, the ACK cannot be assigned to a transaction and gets discarded by if ( is_method("ACK") ) { xlog(,"L_INFO", "WITHINDLG ACK - not loose route\n"); if ( t_check_trans() ) { xlog(,"L_INFO", "WITHINDLG ACK - t_check_trans() \n"); # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { xlog(,"L_INFO", "WITHINDLG ACK - not t_check_trans() DISCARD!!\n"); # ACK without matching transaction ... ignore and discard route(NATMANAGE); #t_relay(); #exit; Any idea? Problem with modifying the sip tags? Or problem with the dialog? Thanks for helping OIi

3 6

Kamailio Iterate All Headers
by Colin Morelli 01 Jul '16

01 Jul '16

Hey all, Back with more questions. I'm using Kamailio to make an HTTP call to my API to perform authentication and message routing. Currently, I'm trying to build up the post body that I send to my API to make those decisions. I've cherry picked a few of the headers that are important in my routing decisions. But, ideally, I'd like to just iterate over all the SIP message headers and append them as request parameters to my API call. Is this possible with Kamailio? I've been looking through the docs and can't seem to find a function to iterate the full list of headers. Thanks in advance. Best, Colin

4 6

Get registered user list
by Никита Лебедев 01 Jul '16

01 Jul '16

Good day! Can anyone point me to right direction? The question is - I want to get list of registered users (to find out who is online at the moment) from sip server. I'm using kamailio on server side and exosip library on client side. I tried to find something in RFC 3261 and failured... Can someone help and tell what to do or where to look for the answer?

3 3

P-Asserted-Identity - Pseudo-Variables are set as null
by José Seabra 30 Jun '16

30 Jun '16

Hello there, The Pseudo-Variablest hat store information about P-Asserted-Identity are all as set null: $pU=<null> $pd=<null> $pn=<null> Is there any known reason for that? The kamailio version that I'm using this 4.4.1 Thank you for you support Best regards -- José Seabra

3 3

Kamailio Load Balancer Architecture
by Daryn Johnson 30 Jun '16

30 Jun '16

I am interested in opinions and suggestions about load balancing with Kamailio. I work for an ITSP that currently uses Oracle & Broadsoft, and I am working to design and develop an open source solution using Kamailio (Proxy, Registrar, LB) & other Application/Media servers for more flexibility and freedom :) Thank you ALL for the work you have put in on Kamailio! After much reading and configuration I have a Kamailio 'proxy' setup, with endpoints registered using the Registrar module, and calls being sent/received to/from the PSTN. I am interested in separating the Load Balancers from the Registrars & logic, for security and in order to be able to scale appropriately. I will be using the dispatcher module for both load balancers and proxies using the following architecture: PublicIP = 5.5.5.5; Private IP = 192.168.1.0/24 USERS (Public Internet) ==> (public: 5.5.5.5) [ Kamailio (LoadBalancer, Firewall, Sanity Checks) ] (core:192.168.1.2) ==> [ Kamailio Registrar, Proxy, PSTN GW ] ==> AppServers or PSTN GW (we can access our PSTN gateways Via our Core using Private IPs) Questions: 1) Is it overkill to separate the LB & Proxy/Registrars? 2) Is this a common architecture & anyone configured this architecture successfully? Thanks in advance for your help! *Daryn Johnson* *Senior VoIP Engineer*

4 8

Kamailio v4.3.6 Released
by Daniel-Constantin Mierla 30 Jun '16

30 Jun '16

Hello, Kamailio SIP Server v4.3.6 stable release is out. This is a maintenance release of the previous stable branch, 4.3, that includes fixes since release of v4.3.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.6 (or to 4.4.x series). For more details about version 4.3.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/06/kamailio-v4-3-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.4, at this moment with its latest release v4.4.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

1 0

Releasing Kamailio v4.3.6
by Daniel-Constantin Mierla 30 Jun '16

30 Jun '16

Hello, as announced last week, I am planning to release a patch update from branch 4.3, respectively v4.3.6, tomorrow. Today I backported what I found to be relevant from branch 4.4 to 4.3. If anyone notices something missing, reply on the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

ipsec support in PCSCF
by Vikram Chhibber 30 Jun '16

30 Jun '16

Hi All, I did not get any concrete information on whether PCSCF supports transport mode ipsec in Kamailio release 4.4. It is mentioned that it is not supported in release 4.0. Could anyone please let me know the status on this? Thanks, Vikram

3 4

kamailio & redis
by Fabian Pignataro 29 Jun '16

29 Jun '16

Hello Community, I'm trying to read data (hash) stored in redis database, from kamailio script. I read the module documentation ndb_redis: http://www.kamailio.org/docs/modules/4.4.x/modules/ndb_redis.html#idp263007… But I'but I don't understand what I'm doing wrong :/ With someone has had experience this? *My platform:* *root@kamailio-fts:~# cat /etc/issue* *Ubuntu 14.04.4 LTS \n \l* *root@kamailio-fts:~# kamailio -V* *version: kamailio 4.4.1 (x86_64/linux) * *flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB* *poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.* *id: unknown * *compiled with gcc 4.8.2* *My script:* *modparam("ndb_redis", "server", "name=srvN;addr=127.0.0.1;port=6379;db=1")* *modparam("ndb_redis", "init_without_redis", 1)* *if(redis_cmd("srvN", "HMGET usuario:2002 status", "res")) {* *xlog("L_INFO","array size: $redis(res=>size) - type: $redis(res=>type) - value: $redis(res=>value[0]) $redis(res=>value[1]) \n"); * * }* *The output of syslog (I don't see errors): * *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7fc2fd008a30* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [db_res.c:60]: db_free_rows(): freeing rows at 0x7fc2fd007020* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [db_res.c:134]: db_free_result(): freeing result set at 0x7fc2fd008a98* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: db_postgres [km_dbase.c:433]: db_postgres_free_query(): PQclear(0xcc9060) result set* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [socket_info.c:564]: grep_sock_info(): checking if host==us: 13==13 && [172.16.20.219] == [172.16.20.219]* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0x7fc2fcfc09a0* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: registrar [lookup.c:240]: lookup_helper(): contact for [2004] found by address* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: INFO: <script>: es un mensaje y estoy por enviarlo* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: INFO: <script>: array size: 2 - type: 2 - value: <null>* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=4 , global msg id=4 , T on entrance=(nil)* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=36630, isACK=0* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_hooks.c:358]: run_reqin_callbacks_internal(): DBG: trans=0x7fc2ed762d08, callback type 1, id 0 entered* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: d79552e0dafc752469a90c3599ba2d42* *Jun 27 11:48:12 kamailio-fts /usr/sbin/kamailio[31615]: DEBUG: tm [t_funcs.c:368]: t_relay_to(): SER: new transaction fwd'ed* It can be seen that $redis(res=>size) & $redis(res=>type) return values, not so with $redis(res=>value[0]). *By last, I share de output of redis CLI:* *127.0.0.1:6379 <http://127.0.0.1:6379>> hget usuario:2002 status * *"Online" * Cheers, Fabián - -- Ing. Fabián Pignataro

2 4

rtpengine allow_op problem
by david 29 Jun '16

29 Jun '16

hello all i have been taking a look to the rtpengine.c code and i found that the issue has relation with the queried_nodes_limit which had 50 defalut value, so before it was reached in the loop, we received a request/response retransmision doing all the same. anycase, the issue it's because when allow_op is applied, the select_rtpp_node gets a node even having the node disabled by the first attemp (having the rtpengine node down), and it does the loop. if it's not applied, we dont get a node at the retry. but adjusting the number of queried_nodes_limit to a value to be reached before a request retransmission arrives, i guess it should be enough (¿?) best regards david escartin

2 2

【Issue of kamailio with pgpool】
by Noriyuki Hayashi 29 Jun '16

29 Jun '16

Hi, I am testing on CentOS 6.6 x86.64 kamailio-4.3.5 postgresql-9.4.6 pgpool-II-pg94-3.5.2 When I using floating IP as 192.168.192.143 that kamailio is working. But I could not access to psql -h 192.168.192.143 -p 9999 -U postgres asterisk. Let me know if you have this issue with much appriciated. Thank you. #!define DBURL "postgres://asterisk:password@192.168.192.143:9999/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "postgres://asterisk:password@192.168.192.143:9999/asterisk" #!endif #!endif Noriyuki Hayashi Japan.

2 2

rtpengine allow_op problem
by david 29 Jun '16

29 Jun '16

hello all i have observed an issue using this parameter, which is useful in case you deactivate a node from the rtpengine sets, to let the calls to finish the negotiation. but i have tested that if a node fails down in the middle of a call, with that option activated, the kamailio gets blocked waiting until the rtpengine node is again up, to finish handling with the BYE request for instance. Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> could it be possible to make the kamailio leave that blocking state, or at least be able to activate the allow_op param by kamctl fifo? thanks a lot and regards david escartin

1 0

Documentation DMQ module
by Jan Hazenberg 29 Jun '16

29 Jun '16

Hi, Found a error in the documentation of the DMQ module (http://kamailio.org/docs/modules/devel/modules/dmq.html#dmq.p.num_workers). In the documentation there is a example: modparam("dmq", "num_threads", 4) If i'm correct this should be: modparam("dmq", "num_workers", 4) Thanks, Jan

2 1

rtpengine allow_op issue
by david escartin 28 Jun '16

28 Jun '16

hello all i have observed an issue using this parameter, which is useful in case you deactivate a node from the rtpengine sets, to let the calls to finish the negotiation. but i have tested that if a node fails down in the middle of a call, with that option activated, the kamailio gets blocked waiting until the rtpengine node is again up, to finish handling with the BYE request for instance. Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:50 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:51 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:52 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:53 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:54 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:55 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:56 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26571]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:57 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:58 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:19:59 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:00 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:01 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:02 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:03 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26573]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26571]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26568]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:04 /usr/local/kamailio/sbin/kamailio[26569]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: BYE received in A-Leg, 106406 Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: INFO: test Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying rtpproxy session on BYE Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: ERROR: rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command "delete" to RTP proxy <udp:x.x.x.x:7772> Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26572]: ERROR: rtpengine [rtpengine.c:2653]: send_rtpp_command(): timeout waiting reply from RTP proxy <udp:x.x.x.x:7772> could it be possible to make the kamailio leave that blocking state, or at least be able to activate the allow_op param by kamctl fifo? thanks a lot and regards david escartin

1 0

Kamailio - PSTN (kamailio) - rtpproxy issue
by Filippo Graziola 28 Jun '16

28 Jun '16

Hello everybody, I have a local kamailio+rtpproxy under NAT listening on multiple local interfaces (vlans) which works fluenty for all local user indipendently and engages rtpproxy correctly. I have another working kamailio+rtpproxy which is accessible from everywhere at a public address and I would like to route to it some specific calls from the local server. So, in the local kamailio configuration file I defined WITH_PSTN (all the necessary option) and INVITE/OK/BYE messages are going correctly to the destination. The problem is that I have no audio on both ways. After some packet sniffing, I tried killing local rtpproxy process and now calls to PSTN have also audio both ways. >From these tests, if my configuration is correct, when routing the call to PSTN gateway I should disable the local rtpproxy, and I'm in doubt about how I can do this. Could you help me? Should I change NAT flags in the PSTN route? Thanks

1 2

Kamailio v4.4.2 Released
by Daniel-Constantin Mierla 28 Jun '16

28 Jun '16

Hello, Kamailio SIP Server v4.4.2 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.2. For more details about version 4.4.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/06/kamailio-v4-4-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Enjoy the summer holidays! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

acc time mode 2 wrong timestamp
by pars3c 28 Jun '16

28 Jun '16

Hello, i've installled the new version of kamailio 4.4.1 . in 2 different server , with acc module and these parameters: modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "seconds") should be seconds.millisec On the first server in acc table , in field seconds i have values like this 1466493457.85, this is correct timestamp ( this server have centos 86_64) In the other server , i have values like 1913113.702 , that is wrong timestamp . (this is old centos 32 bit ) What can be the problem? The different version 32,64 bit ? or some compiler option ? Thanks

2 11

Planning release of v4.4.2 and other v4.x.y
by Daniel-Constantin Mierla 28 Jun '16

28 Jun '16

Hello, I am considering to release kamailio v4.4.2 next week along with the corresponding patch releases for branches 4.3 and 4.2. It will likely be the last release in series 4.2 to mark its end of official packaging. I would also consider a new patch release in 4.1.x, if that is still used by many out there to get packages that fix the libssl version matching -- Victor should say if the packages for this branch can still be generated in order to make sense doing this release. Anyhow, report any issues you are aware and are not yet announced for branches 4.4 and 4.3 to the bug tracker as soon as possible to get a chance to be fixed by release date. Likely 4.4.2 will be next Tuesday, June 29, with the other branches done in the following days. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

userblacklist use avp for check_blacklist
by Jan Hazenberg 28 Jun '16

28 Jun '16

Hello, I'm trying to use the userblacklist module to implement a global blacklist. I have a few questions about the module. Is it possible to use a avp for the check_blacklist function? If i'm correct it will only parse the request URI, but i'm rewriting the URI in a branch route to allow different formats to different destinations. Or is this only possible with the check_user_blacklist function? I also have a issue with numbers in E164 format with a leading +. I tried to escape this with a \ in the database but it will not load the entry's when i use the + in front of the number. Is the + supported by this module or should i strip it before the function and add it again afterwards? Thanks, Jan

2 12

cgrates - cdrs_tables.sql error
by Infinicalls Infinicalls 27 Jun '16

27 Jun '16

Hi, I am trying to install cgrates , but face this issue with database. " ERROR 1067 (42000) at line 7: Invalid default value for 'deleted_at' " is encountered in create_cdrs_tables.sql . Can somebody let me know how to overcome this? regards Ganesh Kumar

2 1

Re: [SR-Users] S-CSCF (kamailio) crashes after deregistration of client
by Christoph Valentin 27 Jun '16

27 Jun '16

2 1

behaviour of uac_redirect module
by Dirk Teurlings - Signet B.V. 27 Jun '16

27 Jun '16

Hi, Testing with latest stable 4.4 Kamailio at the moment, and running into the following issue. I have a setup that sends a 302 to do a Moved Temporarily to another extension. To update the Contact header accordingly, I use the get_redirects function in the failure_route. get_redirects('*', 'Redirects') This results in the following packet U 192.168.10.246:5060 -> 192.168.10.245:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 192.168.10.245:5060;branch=z9hG4bK40b58e09;rport=5060;received=192.168.10.245. From: "+31........" <sip:+31........@192.168.10.245>;tag=as42745abe. To: <sip:+31........@192.168.10.246>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f67d. Call-ID: 576349c40a7cb77a06230e181831672d@192.168.10.245:5060. CSeq: 102 INVITE. Contact: <sip:06........@sip.example.net>, <sip:06........@sip.example.net>;q=0.01. Server: sip.example.net. Content-Length: 0. . This works, but I don't want ACC to account for anything, so I tried: get_redirects('*') But this results in the following packet U 192.168.10.246:5060 -> 192.168.10.245:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 192.168.10.245:5060;branch=z9hG4bK185f4fdd;rport=5060;received=192.168.10.245. From: "+31........" <sip:+31........@192.168.10.245>;tag=as13f23689. To: <sip:+31........@192.168.10.246>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-b5fe. Call-ID: 336a16912db412d418f7f7495e2574f5@192.168.10.245:5060. CSeq: 102 INVITE. Contact: <sip:accountname@sip.example.net>, <sip:06........@sip.example.net>;q=0.01, <sip:06........@sip.example.net>;q=0.01. Server: sip.example.net. Content-Length: 0. . Notice the extra accountname Contact, this results in a forward that doesn't work for the mediaserver. To me the documentation about the function states that there should be no difference between using it with or without accounting as far as SIP messages go. Is this a bug, or is this by design? Cheers, Dirk

2 4

Question about gzcompress module
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, When using the gzcompress module, is it possible to compress/decompress only one specific header, or will it compress/decompress the whole body? Thanks, Joel.

2 2

Get full body from http POST request
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, I'm using http_query (POST method) to interact with an API. This API responds with a JSON. Is there a way to get the full body and after parse that JSON? Example response: { "error": { "code": 404, "message": "Destination not Found" } } Currently, I'm only being able to get the first line, in the previous example it would be: "{" According to the doc, that is the expected behavior: If HTTP server returns a class 2xx, 3xx or 4xx reply, the first line of the reply's body (if any) is stored in “result” parameter, which must be a writable pseudo variable. I don't have the ability to get the response in 1 single line like: { "error": { "code": 404, "message": "Destination not Found" } } How can I get the full body instead of only the first line? Thanks. Best regards, Joel.

1 1

Parsing diversion headers correctly
by Joel Serrano | VOZELIA 26 Jun '16

26 Jun '16

Hi, If you receive an INVITE with 2 diversion headers such as (in this order): Diversion: <sip:9975@X.X.X.X:5060>;privacy=off;screen=no; reason=user-busy; counter=1 Diversion: <sip:1234567890@Z.Z.Z.Z:5060>;privacy=off;screen=no; reason=unknown; counter=1 How is a nice way of extracting the user part from the URI of the last one? (in this case: 1234567890) Problems I have found: $(di{uri.user}) ---> Give mes the user part of the URI of the first header, in this case: 9975 Is there a way to tell $di to get its value from the last header instead of the first one? This example works: $(hdr(Diversion)[-1]{param.name,0}{s.strip,1}{s.striptail,1}{uri.user}) But I think it is very hacky... Any better approach anyone can suggest me? Thanks in advance. Best regards, Joel.

2 2

Public vs Internal IPs
by Colin Morelli 25 Jun '16

25 Jun '16

Hey all, When using Kamailio at the edge - what's the best practice around how to advertise your Record-Route? I assume it's going to involve the use of a double-RR with both the public and private IPs. However, I'm running in AWS where the host doesn't have two interfaces with both a public and private IP. So, I set the host's public IP as the advertised address while Kamailio binds to the local IP. But now how do I get Kamailio to know that it needs to double-RR in this case? Thanks, Colin

2 9

Specify alternative OpenSSL engine for Kamailio
by Joel Serrano | VOZELIA 25 Jun '16

25 Jun '16

Hi, This is a similar question on a previous thread: "[SR-Users] Compiling kamailio with custom openssl" Is it possible to specify a different OpenSSL engine for kamailio to use? For example: On nginx you have the config param: ssl_engine <engine> On apache you have the config param: SSLCryptoDevice <engine> Is there anything similar on Kamailio? We are using debian 8 with openssl-1.0.1t and these are the available engines: root@debian:/usr/lib/x86_64-linux-gnu/openssl-1.0.0/engines# ls -l total 404 -rw-r--r-- 1 root root 19512 Jun 21 11:07 lib4758cca.so -rw-r--r-- 1 root root 19784 Jun 21 11:07 libaep.so -rw-r--r-- 1 root root 15576 Jun 21 11:07 libatalla.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libcapi.so -rw-r--r-- 1 root root 24232 Jun 21 11:07 libchil.so -rw-r--r-- 1 root root 19864 Jun 21 11:07 libcswift.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libgmp.so -rw-r--r-- 1 root root 93304 Jun 21 11:07 libgost.so -rw-r--r-- 1 root root 15432 Jun 21 11:07 libnuron.so -rw-r--r-- 1 root root 6104 Jun 21 11:07 libpadlock.so -rw-r--r-- 1 root root 24096 Jun 21 11:07 libsureware.so -rw-r--r-- 1 root root 19784 Jun 21 11:07 libubsec.so root@debian:/usr/lib/x86_64-linux-gnu/openssl-1.0.0/engines# The reason is we have another server with a Cavium SSL card, which provides a dynamic engine: -rw-r--r-- 1 root root 127968 Jun 23 15:18 libcavium.so If it is not possible via config parameter, is there a way to specify the openssl engine to use before compiling? Thanks in advance. Best regards, Joel.

2 2

Count number of headers of type
by Joel Serrano | VOZELIA 25 Jun '16

25 Jun '16

Hi, It there an easy way to count the number of existing headers of certain type? For example: - How many 'via' headers: $hdrcount(Via) ? - How many 'diversion' headers: $hdrcount(Diversion) ? The $hdrcount is an example, just wondering if there is an easy way to get the number. Thanks. Best regards, Joel.

2 2

t_on_failure route not being called
by Colin Morelli 24 Jun '16

24 Jun '16

Hey all, I'm using a combination of http_async_client and rtjson to query my API and retrieve a JSON target route set for an incoming SIP request that Kamailio will forward to. The HTTP portion of it works great. I'm able to hit my API and get back a JSON document. Additionally, the rtjson parsing appears to work, because the request is hitting my freeswitch instance. However, FS is responding with a 503 (I'm intentionally doing this for testing), and Kamailio isn't calling my t_on_failure route. I've pasted the relevant portion of my config below - would be great if anyone has any insights. tl;dr: the "REROUTE" route section below isn't being reached even when FS responds with a 503. Best, Colin route[ROUTING_REPLY] { if $http_ok && $http_rs == 200 { xlog("L_INFO", "$ci|log|loaded target route set from document $http_rb\n"); rtjson_init_routes("$http_rb"); rtjson_push_routes(); xlog("L_INFO", "$ci|log|attempting to relay request towards $du\n"); t_on_failure("REROUTE"); t_set_fr(0, 1000); t_relay(); } else { send_reply("480", "Temporarily Unavailable"); } } route[REROUTE] { xlog("L_WARN", "$ci|log|relay to destination $du failed, checking next route\n"); if (rtjson_next_route()) { xlog("L_INFO", "$ci|log|additional route found, directing request to $du\n"); t_on_failure("REROUTE"); t_set_fr(0, 1000); t_relay(); } else { xlog("L_WARN", "$ci|log|route list exhausted, failing request\n"); send_reply("480", "Temporarily Unavailable"); exit; } }

1 1

S-CSCF (kamailio) crashes after deregistration of client
by Christoph Valentin 24 Jun '16

24 Jun '16

4 12

Kamailio behind ELB AWS
by yann christophe 24 Jun '16

24 Jun '16

Hello Everyone, I would like to know, if i can use ELB and route53 behind some Kamailio servers for the high availibility ? I use Kamailio behind Freeswitch. I would like deploy these servers on AWS. What do you suggest me for the high availibility on AWS with Kamailio. Thanks Best regards Yann

2 1

rtpengine installation delete ethernet interface on Centos
by Yasin CANER 24 Jun '16

24 Jun '16

1 0

How to understad that user is de-registrating
by Alexandru Covalschi 24 Jun '16

24 Jun '16

Hello list, I need to send to an external API events when user is registrated and de-registrated. As far as I understand standart behaviour is as follows: If user is not registered, he sends REGISTER and he is registrated (I can catch that because I make the auth). If user is registered and sends REGISTER he is de-register. (Please correct me if I'm wrong.) How can I catch that? Can I use event_route[usrloc:contact-expired]? Thanks in advance! -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator tel: +37367398493 web: http://abriss.solutions/ <http://abs-telecom.com/>

3 4

uacreg flags field not changing
by Yuriy Gorlichenko 24 Jun '16

24 Jun '16

Hi. Im using kamailio 4.4.1 kamailio -v flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 99e1c3 compiled on 13:34:38 Jun 14 2016 with gcc 4.9.2 I using uacreg table for register trunks and want to know their status For now i see status onlu from kamcmd uac.reg_dump/info But if i trying to take flags from uacreg table (flags field) it always 0.

1 0

Compiling kamailio with custom openssl
by Cibin Paul 24 Jun '16

24 Jun '16

Hi, Can any one guide me to compile kamailio with custom openssl path please. Thanks & Regards Cibin

2 12

Dialog timeout before answer
by Alex Balashov 24 Jun '16

24 Jun '16

Hi, Will the dialog timeout - and specifically, $dlg_ctx(timeout_bye) - also apply to dialogs which have not yet been established? If so, does that mean the dialog module will send a CANCEL instead for a dialog in an early/progress state? -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

kamailio + fax "other fax not responding"
by Mx R 23 Jun '16

23 Jun '16

Hi there, We use kamailio 4.3.3. And we have one problem with the fax. We send and receive from and to our network, the fax does not work. Fax reports "other fax not responding". When we send through our network to another operator they receive fax. What could be wrong?

2 5

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 23 Jun '16

23 Jun '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0

3 5

Dialog in delete state is too old
by Emilio Soler Muñoz 23 Jun '16

23 Jun '16

Hello! Kamailio process has stopped. Anyone else has happened? Versión: 4.4 This is what I have in the logs. kamailio.log: 2016-06-23T10:10:33.983494+02:00 kam01 /usr/sbin/kamailio[2907]: NOTICE: dialog [dlg_hash.c:245]: dlg_clean_run(): dialog in delete state is too old (0x7f092e15b190 ref 3) 2016-06-23T10:10:34.005015+02:00 kam01 /usr/sbin/kamailio[2897]: WARNING: dialog [dlg_handlers.c:1348]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7f092e15b190 [3554:5262] with clid 'L, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PU' and tags 'as6b8e8af9' 'as25960d84' 2016-06-23T10:10:34.007989+02:00 kam01 /usr/sbin/kamailio[2913]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 2016-06-23T10:10:34.008119+02:00 kam01 /usr/sbin/kamailio[2823]: ALERT: <core> [main.c:739]: handle_sigs(): child process 2897 exited by a signal 11 2016-06-23T10:10:34.008128+02:00 kam01 /usr/sbin/kamailio[2823]: ALERT: <core> [main.c:742]: handle_sigs(): core was not generated kern.log: [363406.951756] kamailio[2897] general protection ip:7f092c9413a5 sp:7ffdd1065530 error:0 in dialog.so -- Emilio Soler Muñoz <emiliosoler(a)gmail.com>

1 0

Add a user through a http request
by Jay Li 22 Jun '16

22 Jun '16

Dear All, I wonder if Kamailio has the functionality to add a subscriber dynamically. For example, if I use event_route[xhttp:request] to expect a http get request, then based on the received request, add a user in the DB. Not sure what the best approach would be. Any suggestion is appreciated. Thanks. Regards,Jay

2 1

Dispatcher - Route call to alternate server if there is delay to answered call
by Amit Patkar 22 Jun '16

22 Jun '16

Hi I have a scenario where dispatcher is used to balance calls. Assume there are 2 media servers M1 & M2. Call are routed using round robin algorithm. There is scenario where call is handed over to M1 but it is not answered in defined time period (say 5 seconds due to load on server or network congestion), I wish to send this call to M2. What are the options to achieve it? Thanks & Regards, Amit Patkar

2 1

Re: [SR-Users] uac.reg_refresh not removing records from memory
by DanB 22 Jun '16

22 Jun '16

Hey Daniel, Thanks for so fast feedback! No difference as far as I can tell before and after removing the record from db. I have also not disabled specifically the record inside uacreg table but completely removed it from db. I should mention that I have just upgraded to latest 4.4.1 so I should have the flags also saved in db. Here is the record before removal: """ root@iPBXDev1:/etc/ipbxrp# kamcmd -s tcp:127.0.0.1:2046 uac.reg_dump { l_uuid: rsua1_172.16.254.101 l_username: rsua1 l_domain: 172.16.254.101 r_username: cust_sua1 r_domain: 172.16.254.102 realm: auth_username: cust_sua1 auth_password: check123 auth_proxy: sip:172.16.254.102:5060 expires: 3600 flags: 20 diff_expires: 3588 timer_expires: 1466349066 reg_init: 1466345451 reg_delay: 0 } """ and the one after: """ root@iPBXDev1:/etc/ipbxrp# kamcmd -s tcp:127.0.0.1:2046 uac.reg_dump { l_uuid: rsua1_172.16.254.101 l_username: rsua1 l_domain: 172.16.254.101 r_username: cust_sua1 r_domain: 172.16.254.102 realm: auth_username: cust_sua1 auth_password: check123 auth_proxy: sip:172.16.254.102:5060 expires: 3600 flags: 20 diff_expires: 3526 timer_expires: 1466349066 reg_init: 1466345451 reg_delay: 0 } """ Also here is the command refreshing the record (after that is not longer in db): """ POST /jsonrpc HTTP/1.1. Host: 172.16.254.101:5090. User-Agent: Go-http-client/1.1. Content-Length: 85. Content-Type: application/json. Accept-Encoding: gzip. . {"jsonrpc":"2.0","method":"uac.reg_refresh","params":["rsua1_172.16.254.101"],"id":7} # T 2016/06/19 16:17:47.275473 172.16.254.101:5090 -> 172.16.254.102:58172 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP 172.16.254.102:58172. Content-Type: application/json. Server: Sipean-RP 4.3.2. Content-Length: 36. . {"jsonrpc":"2.0","result":{},"id":7} """ Ta, DanB On 16.06.2016 12:00, sr-users-request(a)lists.sip-router.org wrote: > Message: 10 Date: Thu, 16 Jun 2016 08:17:53 +0200 From: > Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: > [SR-Users] uac.reg_refresh not removing records from memory > Message-ID: <76d9f2eb-26f7-c0db-8b74-37c93a5d3ef4(a)gmail.com> > Content-Type: text/plain; charset=utf-8 Hello, On 14/06/16 16:52, DanB > wrote: >> > >> >Hey Guys, >> > >> >I was wondering if there is any dynamic way today to remove a single >> >registration out of memory (and implicitly to force un-REGISTER) >> > >> >Eg: I add one record using uac.reg_refresh (which works fine) but after >> >removing the record from the database and executing uac.reg_refresh >> >again, the record stays in memory until expire. >> >Is there a better way to enforce un-register on remote side? >> > > do the flags in memory of the uac_reg account show that it is disabled? > > Cheers, > Daniel

4 3

WS to WS calls - No ACK received for 200OK
by Amit Patkar 21 Jun '16

21 Jun '16

Hi I am using Kamailio as Websocket proxy. User 1 & User 2 are registered on Kamailio over WebSocket. When User 1 calls User 2, User 2 gets ring and answers the call. 200 OK message is received by User 1 but ACK response sent by User 1 does not reach User 2. Since User 2 didn't get ACS, after 30 sec timeout it drops the call. Media is exchanged for 30 seconds, which means ICE is successful. I am running kamailio 4.3.3 on Kamailio is behind firewall and running on private IP. rtpengine is configured to handle media. All ports are forwarded to kamailio server and Kamailio is allowed to send data on any public IP. I used jssip & sipml framework to test. Result was same for both frameworks. No ACK received I can see following errors in log file. What could be the reason which indicate ACK was not forwarded Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6584]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: transaction answered: timestamp=1466501141;method=INVITE;from_tag=7926B9NR7U0X99ZfAr1T;to_tag=w3D2JGd2EUFJpobTcFyo;call_id=1edc8e15-f1f1-584d-df81-71c3d59d713d;code=200;reason=OK;src_user=10001;src_domain=xxxx.com;src_ip=xxx.yyy.zzz.aaa;dst_ouser=10002;dst_user=10002;dst_domain=df7jal23ls0d.invalid Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6584]: INFO: <script>: WS to WS Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:46 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:46 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:51 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:51 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:55:57 acstemplate /usr/sbin/kamailio[6582]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:55:57 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) Jun 21 14:56:13 acstemplate /usr/sbin/kamailio[6581]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket Jun 21 14:56:13 acstemplate /usr/sbin/kamailio[6581]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) I searched list for similar issues but could not find match any Help to resolve this issue is appreciated. Thanks & Regards, Amit Patkar

3 4

Kamailio consult CNAM server through SIP Subscribe.
by José Seabra 21 Jun '16

21 Jun '16

Hello there, I need to use kamailio to consult an CNAM server in order to get the caller id name and number to the call, the diagram bellow represents what i need implement: Can I implement this behavior through the following kamailio modules: 1. tm and tmx module (call the function t_suspend and t_continue in order to stop INVITE transaction and resume it). 2. UAC module (generate the SUBSCRIBE sip message) 3. Some kamailio scripting logic to manage this behavior. what are your advices to implement this scenario? Thank you for your help Regards -- Cumprimentos José Seabra

4 17

kamailio question
by Irakli Nanobashvili 21 Jun '16

21 Jun '16

dear sr_users i have a little problem about kamailio sip server and who can help me ?

2 1

Equivalent of OpenSIP's binary interface in Kamailio
by Gholamreza Sabery 21 Jun '16

21 Jun '16

In OpenSIPS binary interface ( https://www.opensips.org/Documentation/Interface-Binary-2-2) provides a more efficient way of replicating state between multiple OpenSIPs instances. What is the equivalent feature in Kamailio? Regards

2 2

RTPENGINE rtp errors
by Dmitry 21 Jun '16

21 Jun '16

Hi 1)I encountered such errors when I make an internal call between 2 yate client SIP softphones:rtpengine is used and no audio: | Jun 19 10:02:35 kazootest2 rtpengine[2066]: [918508371(a)kazootest2.siplabs.local port 30942] Error parsing RTP header: short packet (header) when I turn off the rtpengine:THe Freeswitch handles such RTP packets correctly and audio is ok. I compared manually rtp packets from Yate: they comply to the RTP specification. 2) A call from WS client(jssip) to an ordinary sip softphone: | Jun 19 09:58:36 kazootest2 rtpengine[2066]: [s2m2k70tu0d12bjnnamr port 30814] Error parsing RTP header: invalid header version | How to fix these errors? The rtpengine looks through all incoming packets while looking for a new RTP stream. May be packets parsing is too strict? |

1 1

No other running SIP apllication on the machine.//答复: Kamailio ERROR "Address already in use" in tcp_init().
by joey.ren＠yulong.com 20 Jun '16

20 Jun '16

2 1

Kamailio dispatcher module - keep alives
by José Seabra 20 Jun '16

20 Jun '16

Hello there, I have a Kamailio server using dispatcher module to monitor several peers, some of them are in the same network, others are in an external network, for the peers that are in the external network kamailio has to send the OPTIONS(dispatcher keep alives) through another kamailio located in the border and acting as loadbalancer. If i set the dispatcher option "outbound_proxy" all SIP OPTIONS are sent to the outbound proxy configured and it isn't what i need, I need to have by some way a possibility to set outbound_proxy only for the peers that are located on the external network. How can I do that? Regards -- Cumprimentos José Seabra

3 5

Kamailio ERROR "Address already in use" in tcp_init().
by joey.ren＠yulong.com 20 Jun '16

20 Jun '16

3 2

Why SDPOPS does not remove attributes in SDP
by Dmitry 18 Jun '16

18 Jun '16

Hi allI have the following code: if($T_reply_code=="200") { if(has_body("application/sdp")) { xlog("L_INFO", "RTPENGINE received internal reply $T_reply_code $rr SDP extra lines will be removed"); set_rtpengine_set("0"); rtpengine_manage(); sdp_remove_line_by_prefix("a=rtcp"); sdp_remove_line_by_prefix("a=ssrc"); sdp_remove_line_by_prefix("a=ice"); sdp_remove_line_by_prefix("a=candidate"); xlog("L_INFO", "RTPENGINE received internal reply $T_reply_code $rr SDP extra lines removed with SDPOPS"); } }When I look through traces - I see that 200 ok(with SDP) has all these attributes and they are not removed. Why SDPOPS does not remove these attributes?

4 5

Help with routing block and Dispatcher Module
by pablo rosales 17 Jun '16

17 Jun '16

Hi everyone! I am a newbie with Kamailio. I'm trying to connect a kamailio server to an asterisk gateway. We are using basic Kamailio 4.4.x configuration file with mysql and rtpproxy we are having same issues with dispatcher module. we can not call between local subscribers. Basically, If I call to a local subscriber, the module always execute the route(DISPATCHER) code on the routing block, trying to send the call to my gateway, but if I try to call to an external number, it works perfect. All calls are being send to my gateway. Here is my dispatcher code: # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } ### xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); xlog(" --- DISPATCH: going to <$ru> via <$du>n"); t_on_failure("RTF_DISPATCH"); return; } ## Sample failure route failure_route[RTF_DISPATCH] { xlog(" --- DISPATCH: Starts RTF_DISPATCH \n"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } Can anybody help to see what I am doing wrong??

2 2

http_client module configuring User-Agent
by Jurijs Ivolga 17 Jun '16

17 Jun '16

Hi All, I tried today new module http_client. I just made GET request using http_client_query in following way: http_client_query("http://cool.api.com", "$var(result)"); It is working, but I can't see User-Agent in request. Below you can find my module options: modparam("http_client", "useragent", "kamailio") modparam("http_client", "connection_timeout", 1) Is it a bug, or am I doing something wrong? Thank you! Jurijs

2 3

Kamailio Problem carrying media
by Sirvan Paraste 16 Jun '16

16 Jun '16

Hi there We have got a situation that needed your help to overcome that. We have setup a Kamailio with voice, video call and text messaging and even push notification ability. it was working and till these days that we have lost media (voice and video and calling ability). seems that the clients do not see each other but strange is that they can send text messages and even push notification is work as well. Could you please give me an idea how to find the problem, or if you know how to solve it? Best regards. Sirvan Parasteh

2 1

Kamailio on multiple ports
by huseyin kalyoncu 16 Jun '16

16 Jun '16

Hello, We have kamailio running as proxy in front of a freeswitch box. This kamailio also acts as a registrar listening both 5060 and 5090 ports. We have two devices registered to kamailio: Device A on port 5060 Device B on port 5090 When Device B calls Device A everything is fine. Call released successfully. When Device A calls Device B, call setup is ok and they can talk. But when Device A tries to release the call, Device B remains open which cannot get a BYE message. When I dug up for the cause of the problem, I realized that: When Kamailo receives 200OK reply on port 5090 for the UPDATE message from Device B, it relays this reply internally and then sends this reply to freeswitch box on port 5060. This results in loosing alias in the contact field of the reply message and then which causes freeswitch to return BYE message with the wrong IP which is the local IP of the Device B. As a result this BYE message cannot be delivered to Device B. Is this a normal behaviour ? If it is so what changes should I make in the kamailio config to fix this? You can find the ngrep outputs of the both calls as attachment. Thanks Huseyin Kalyoncu

1 1

just a quick glance at my kamailio.cfg (suspect a rtpproxy_manage behavior)
by Sébastien Brice 16 Jun '16

16 Jun '16

Hi guys i am bridging call on a private ip asterisk behind kamailio/rtpproxy i am calling rtpproxy with the route [FROMASTERISK] route[FROMASTERISK] { if(ds_is_from_list()){ xlog("L_INFO","$dd Call from Media-Server Cluster\n"); rtpproxy_manage("cawei"); SIP INVITE with sdp always report a corrupt body!!! Session Name (s): Asterisk PBX 13.9.1 Connection Information (c): IN IP4 PUBLICIPPUBLICIP Connection Address: PUBLICIPPUBLICIP thats really strange since i am calling rtpproxy_manage only once i can't understand it appears twice. what do you think about that ? thx you

2 2

ACK not sent out
by Nagorny, Dimitry 16 Jun '16

16 Jun '16

Hello List, I have a setup of two WebRTC-Clients registered to kamailio. RegA calls RegB. Signalling itself for connection seems to work until the point RegA receives the 200Ok and wants to send the ACK, kamailio throws errors. Here are the 200Ok with the following ACK and the errors that kamailio writes: 200OK (from RegB to RegA, successful): SIP/2.0 200 OK Record-Route: <sip:zfHrSW4KTMuuxQYK+gVKH5MK+gUR1jc=@10.250.5.74:8083;transport=ws;lr=on;ftag=3bege11omg> Via: SIP/2.0/WSS 192.0.2.1;rport=54839;received=10.250.5.17;branch=z9hG4bK415866 To: <sip:dnagorny@r5dev-kam1.robot5.de>;tag=ecmmlvc4ee From: "Testaccount" <sip:4@r5dev-kam1.robot5.de>;tag=3bege11omg Call-ID: vu8rkkgki2uupuf2cqf4 CSeq: 2017 INVITE Contact: <sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;alias=10.250.5.17~54838~6;transport=ws> Supported: ice,replaces,outbound Content-Type: application/sdp Content-Length: 1001 v=0 o=mozilla...THIS_IS_SDPARTA-47.0 4652302965311172452 0 IN IP4 0.0.0.0 s=- t=0 0 a=sendrecv a=fingerprint:sha-256 1A:88:2D:6E:9F:82:14:E0:A2:DB:A7:16:56:00:2C:25:1B:D2:D1:EE:B2:D5:ED:6F:E1:7C:5C:B4:1A:30:9C:85 a=group:BUNDLE audio a=ice-options:trickle a=msid-semantic:WMS * m=audio 30710 UDP/TLS/RTP/SAVPF 111 c=IN IP4 10.250.5.74 a=candidate:0 1 UDP 2122252543 10.250.5.17 52581 typ host a=candidate:1 1 UDP 1686052863 87.158.246.174 49807 typ srflx raddr 10.250.5.17 rport 52581 a=end-of-candidates a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=fmtp:111 maxplaybackrate=48000;stereo=1 a=ice-pwd:4d722747e7d3b3caa360ebe9e646fb44 a=ice-ufrag:2b7e23b8 a=mid:audio a=msid:{5cea7c38-cc3e-4bd2-9a3b-5d2381f0acf9} {0236e1cf-dfdc-4c94-8ce0-59f52a23ebb1} a=rtpmap:111 opus/48000/2 a=setup:active a=ssrc:565594393 cname:{7df4b0da-093c-4c2d-b154-0b860b43f575} a=sendrecv a=rtcp:30710 a=rtcp-mux a=candidate:Y2XhN5oQNoH6nRph 1 UDP 2122252799 10.250.5.74 30710 typ host ACK (from RegA to RegB, unsuccessfull): ACK sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;transport=ws SIP/2.0 Route: <sip:zfHrSW4KTMuuxQYK+gVKH5MK+gUR1jc=@10.250.5.74:8083;transport=ws;lr=on;ftag=3bege11omg> Via: SIP/2.0/WSS 192.0.2.1;branch=z9hG4bK2821443 Max-Forwards: 69 To: <sip:dnagorny@r5dev-kam1.robot5.de>;tag=ecmmlvc4ee From: "Testaccount" <sip:4@r5dev-kam1.robot5.de>;tag=3bege11omg Call-ID: vu8rkkgki2uupuf2cqf4 CSeq: 2017 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: outbound User-Agent: JsSIP 1.0.1 Content-Length: 0 Thrown Errors: kamailio[5009]: INFO: <script>: Is natted kamailio[5009]: INFO: <script>: From: sip:4@r5dev-kam1.robot5.de To: sip:l1jevtog@192.0.2.59;alias=10.250.5.17~54838~6;transport=ws @ <null> Via: SIP/2.0/WSS 192.0.2.1;branch=z9hG4bK4386608 Route: <null> vu8rkkgki2uupuf2cqf4 kamailio[5009]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket kamailio[5009]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) I don't understand these errors. Can anybody explain and point me what needs to be fixed? Using Kamailio 4.3.5 on CentOS7.2, funny info if I call from outside(PSTN/FreeSWITCH) to kamailio into WebRTC-Client it works w/o problems. Respectfully Dimitry Nagorny Trainee

1 1

OpenSIPS balcklist/whitelist related question
by eyatsko 16 Jun '16

16 Jun '16

Hello! Could you, please, help me to make the above?.. I already have rough understanding about blst, permission modules etc. But I am about to compose CFG to count wrong (erroneous) attempts to REGISTER or INVITE from the same IP through several SIP dialogs. For example, some malicious user attempes to brute-force password. I would like to move it permanently into permissions' database after several (predefined) attempts. Kind regards, Ellad

1 0

uac.reg_refresh not removing records from memory
by DanB 16 Jun '16

16 Jun '16

Hey Guys, I was wondering if there is any dynamic way today to remove a single registration out of memory (and implicitly to force un-REGISTER) Eg: I add one record using uac.reg_refresh (which works fine) but after removing the record from the database and executing uac.reg_refresh again, the record stays in memory until expire. Is there a better way to enforce un-register on remote side? Thanks in advance! DanB

2 1

Question about testing RTPENGINE
by Dmitry 16 Jun '16

16 Jun '16

Hello I am trying to make a test setup to test the RTPENGINE. Case 1: Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.136)-----via NAT(10.111.101.3)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) Case 2: When A calls to B - the sound both ways, so it is ok. Subsriber A ---Asterisk (10.20.20.101)-----via NAT(10.11.97.12)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) WITH RTPENGINE: When A calls to B - the sound is absent, no sound. I compared both dumps the difference only in addressesAnd what I read in nathelper module: - 1 - Contact header field is searched for occurrence of RFC1918 addresses. - 2 - the "received" test is used: address in Via is compared against source IP address of signaling - 4 - Top Most VIA is searched for occurrence of RFC1918 addresses - 8 - SDP is searched for occurrence of RFC1918 addresses - 16 - test if the source port is different from the port in Via - 32 - test if the source IP address of signaling is a RFC1918 address All flags can be bitwise combined, the test returns true if any of the tests identified a NAT. in nat-traversal-role.cfgI see: if (nat_uac_test("3")) { force_rport(); fix_nated_contact(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } As I understand - both addresses are RFC1918but in case 2 the RTPENGINE is not kernalized Does the RTPENGINE distinguish the ip address? why it was not kernalized in case 2? Any help is appresiated.

1 1

RTPENGINE not kernalizing in some cases
by Dmitry 16 Jun '16

16 Jun '16

Hi I have the Kamailio and FS behind it (in Kazoo). Both are in one ip address 10.34.101.104 i use rtpengine_manage(). When I call from Asterisk which is behind NAT :1) Asterisk in VMware player (NAT, local ip 192.168.175.136) and public 10.34.101.103 - the voice was ok? in logs I saw Kernalizing media 2)Asterisk in KVM(NAT, local 10.28.40.101 public 10.101.29.45) and no Kernalizing in /var/log/messages and no voice. The Asterisk configs are absolutely the same. Kamailio log shows no errors In both cases the INVITE's SDP connection is 10.34.101.104 - the address of the Kamailioin the ' case 1' I see all RTP streams : 2 rtp streams - between asterisk and kamailio and vice versa, 2 -rtp streams between kamailio/rtpengine - Freeswitch and vice versa in the second case I see the following RTP streams: 1 rtp stream - from Kamailio to Asterisk, 1 rtp stream - from Freeswitch to Kamailio How to debug such cases? Any help is appreciated.

1 1

HTTP query at the end of a phone call
by Jay Li 15 Jun '16

15 Jun '16

Dear All, I'd like to send a HTTP call to a HTTP server whenever a call routed through a Kamailio server ends. By looking at the document, seems I should use http_query() function in the utils module. I wonder which route function I should put the http_query() in assuming we use the default kamailio.cfg. Thanks. Regards,Jay

4 3

404", "Not Found location"
by Sébastien Brice 15 Jun '16

15 Jun '16

Hi guys when i try to register i have route[LOCATION] that returns "404 Not Found location" due to this statement: $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found location"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } thats weird i did follow ubuntu realtime trying to get sip account registered and then uac_req_send to asterisk but the first step already failed. :-( i noticed there is no Username in my RURI (whatever UAC i am using) maybe this is the root of evil please find attache my kamailio.cfg i am using kamailio v 4.4 Any idea very welcome

2 1

Kamailio + ToS
by Marcelo Bonin 15 Jun '16

15 Jun '16

Hello, I need to know what are the available parameters to set ToS on Kamailio. I couldn't found much information about that, only this https://www.kamailio.org/wiki/cookbooks/4.0.x/core#tos but there is no reference for the available parameters. Thanks, Marcelo.

2 1

question about RTPengine
by Dmitry 14 Jun '16

14 Jun '16

hello i read the documentation about RTPengine.and the documentation says that: If INVITE with SDP, when the tm module is loaded, mark transaction with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for rtpengine_answer() what does it mean? actually - according to 3261 RTP media (both ways) may be turned on after ACK

3 5

IP TOS
by Daniel W. Graham 14 Jun '16

14 Jun '16

Could I get some clarification on tos global setting as referenced by documentation below? Specifically, are DSCP values allowed? Or is this simply for scheduling purposes at the OS level? I can change the value once packets hit the network but if possible would like 0x60 to originate from Kamailio. tos The TOS (Type Of Service) to be used for the sent IP packages (both TCP and UDP). Example of usage: tos=IPTOS_LOWDELAY tos=0x10 tos=IPTOS_RELIABILITY Thanks! -Dan

2 3

Fwd: [Free-RTC] GSoC progress, more mentors?
by Daniel Pocock 14 Jun '16

14 Jun '16

This report was sent to the Free-RTC list https://lists.fsfe.org/mailman/listinfo/free-rtc Most of the projects have an emphasis on SIP If anybody wants to facilitate testing the new Telepathy/Empathy SIP connection manager code against Kamailio or other SIP proxies, registrars and presence servers that would be very welcome. -------- Forwarded Message -------- Subject: [Free-RTC] GSoC progress, more mentors? Date: Mon, 13 Jun 2016 22:53:46 +0200 From: Daniel Pocock <daniel(a)pocock.pro> Reply-To: Free real-time-communications discussion list <free-rtc(a)lists.fsfe.org> To: Free-RTC(a)lists.fsfe.org Hi all, The GSoC students have officially been working for 3 weeks now. I created a summary of all the students here: https://wiki.debian.org/SummerOfCode2016/RTCProjects They've been publishing weekly reports on the debian-outreach list: https://lists.debian.org/debian-outreach/ It would be really helpful to increase the size of the mentoring team for the students working on RTC projects: - we probably need at least one more Python developer and one more C++ developer to help - other people can also help test the projects, for example, if you are using Wordpress, please try Mesut's new WPCall module: https://github.com/mesutcang/wpcall As a bonus, some mentors are selected every year for a fully paid trip to California for Google's GSoC mentor summit. If anybody has more questions about it or would like to be more involved please contact me privately. Regards, Daniel _______________________________________________ Free-RTC mailing list Free-RTC(a)lists.fsfe.org https://lists.fsfe.org/mailman/listinfo/free-rtc

1 0

Kamilio as Scalable Registrar + Proxy
by Colin Morelli 13 Jun '16

13 Jun '16

Hello, I currently have Freeswitch acting as a B2BUA, handling registrations, routing, etc for a prototype voice application I have built. Now I'm at the point where it's time to actually scale it out, and I'm looking at a few different options for the SIP proxy + registration later. Essentially, what I would like to have (I think), is: 1) Stateless SIP proxy with consistent hashing to route to: 2) Transaction-stateful SIP proxy handling registrations and initial routing Ideally, the only thing that's stored across the cluster (if possible) would be registrations, in something like memcache/redis. Then, of course, each node might have its own internal state of transactions that it's tracking. I'd prefer the only thing to care about dialog state to be Freeswitch. Finally, the decisions about which FS instance to route to would be made by my application as well, based on runtime rules. Is this something people here have done before? Are there other suggestions on how to scale out kamailio to multiple instances. Thanks in advance, Colin

2 1

BLF lights
by Bill 13 Jun '16

13 Jun '16

I have spent a couple days beating my head against the Kamilio BLF feature. But cannot seem to get presence signals from one ua to another. I am testing with a Zoiper softphone and a Yealink hardware phone, both registering to Kamailio and able to make and receive calls. But getting absoultely nowhere with BLF. Have gone through all the suggestions in whitepaper 43, but still no results. Anybody had success with BLF, please steer me in the ritght direction. Thanks, Bill

2 1

Kamailio benchmark app_python + OPTIONS
by Anonim Stefan 13 Jun '16

13 Jun '16

Hi, I am doing some basic benchmarks with Kamailio. I am using a VM with 1 CPU i7 2,50 GHz and Kamailio with children=1. This is what I measure for each request, inside the benchmark block: - app_python processing (some euclidian distance based on a dlg_var input) -> same for all requests - uac_req_send() OPTIONS -> same for all requests I am running SIPp tests for 1000 calls with 1,10,50,100,150,200 calls per second. What I see is that the Global average processing time per request decreases with the increasing rate; I expected it to increase with the increasing rate. Is there an explanation for this, or am I doing something wrong ?! Thanks, Stefan

3 5

BYE message to an individual client
by kai.ohnacker＠cbc.de 13 Jun '16

13 Jun '16

Hello Community, is it possible to send a BYE message to an individual client via mi command line (siremis)? Cheers, Kai

2 1

TM Module t_suspend question
by gmele 10 Jun '16

10 Jun '16

Hello, we are using the TM Module* t_suspend *function in our kamailio to suspend transactions (mainly INVITE requests). We do this as we send some push notification to the callee that will force it to re-REGISTER. Once we get the REGISTER, we resume the transaction with a *t_continue()*. If the callee never re-REGISTER, the FR Timer will terminate the INVITE transaction. This is working fine for us when the callee has a phone that can receive the push notification. Now, we want to be able to contact users having multiple devices: phone and desktop VoIP clients. The problem with current implementation is that if we wait for the re-REGISTER from the mobile phone, we may never send the INVITE to the desktop VoIP client if the mobile phone is shutdown. So, I'd like to now if it is possible to set a timer to the t_suspend command which will automatically resume the suspended transaction (without issuing an ERROR like the FR timer is doing)? I didn't see such possibility in the TM module, but is there another way to implement this? Hope I was clear enough Regards Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/TM-Module-t-suspend-question-tp1493… Sent from the Users mailing list archive at Nabble.com.

3 6

Re: [SR-Users] Push notification question
by Vasiliy Ganchev 10 Jun '16

10 Jun '16

Hi! To cope with your specific cases try connection-oriented transport (tcp/tls). e.g. if app is closed - the connection is closed as well and the usrloc record will be removed. After this use standard approach (with late forking - tsilo module) imho, it is not a good idea to suspend the call if there is records in location. Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/Push-notification-question-tp149025… Sent from the Users mailing list archive at Nabble.com.

1 0

Handling ReINVITEs
by Grant Bagdasarian 09 Jun '16

09 Jun '16

Hello, How should reINVITEs be handled in the kamailio script and also in combination with rtpengine? I was reading the RR docs and loose_route function pretty much handles this if I'm correct. We're currently receiving reINVITEs for checking if a session is still active, not for session modifications like SDP changes. Does kamailio handle reINVITEs correctly out of the box or should extra logic be implemented? Also, what happens when rtpengine_manage is called for a reINVITE inside WITHNDLG or in onreply_route for 200 OK? Is the module smart enough to ignore port (de-)allocations when the SDP is still the same as the initial INVITE or 200 OK? Regards, Grant

2 2

forward calls from kamailio(RTPengine) to FREESWITCH
by Ankitt Sharma 09 Jun '16

09 Jun '16

hello i am using using kamailio + RTPengine in my setup and i want to forward every single call to freeswitch(for recording purpose), all three are running on the same machine (testing in the local network) my requirement for media flow is: UA1-->RTPproxy-->>freeswitch-->RTPproxy-->UA2 my user agents are using webrtc media profile so my rtp engine flags are : rtpengine_manage("trust-address replace-origin replace-session-connection RTP/SAVPF"); i've attached my configuration file i've configured freeswitch for media_webrtc=true ,still when i try calling the freeswitch send 488 without sending call to other end should media-address flag be used to forward media through, if yes how it should be done ??

2 1

ACK does not match transaction after stripping Route header
by Francisco Valentin Vinagrero 09 Jun '16

09 Jun '16

Hi there, I'm having an issue in a SBC (ACME) -> KAMAILIO -> Asterisk scenario with an ACK that gets ignored in Kamailio because it does not match any transaction. The INVITE coming from the SBC looks like this (only relevant headers and hidden numbers for simplicity - SBC has IP .12 , Kamailio .30 and Asterisk .34) INVITE sip:mynumber@10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip: mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip: a-number @10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:41754112601@10.15.1.12:5060;transport=udp> And its forwarded to Asterisk with the Record-Route header: INVITE sip: mynumber @10.15.1.30:5060 SIP/2.0 *Record-Route: <sip:10.15.1.30;lr=on;ftag=SDkbo9901-42090> *Via: SIP/2.0/UDP 10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 To: <sip:mynumber@10.15.1.30:5060> From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 P-Asserted-Identity: <sip:a-number@10.15.1.12> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 INVITE Contact: <sip:a-number@10.15.1.12:5060;transport=udp> Then, 200 OK from Asterisk: SIP/2.0 200 OK *Via: SIP/2.0/UDP 10.15.1.30;rport=5060;received=10.15.1.30;branch=z9hG4bK1c02.7dc1b94be22d8780df5141f9ba3c5b7b.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a CSeq: 1 INVITE Server: Asterisk PBX 13.8.0 Contact: <sip:10.15.1.34:5060> Which is sent to the SBC like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdd1m7b00aom47rggc700.1 *Record-Route: <sip:10.15.1.30;lr;ftag=SDkbo9901-42090> Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a CSeq: 1 INVITE Server: Asterisk PBX 13.8.0 Contact: <sip:10.15.1.34:5060> And finally the SBC sends the ACK: ACK sip:10.15.1.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.15.1.12:5060;branch=z9hG4bKdt7p9k00dounet8ic600.1 To: <sip:mynumber@10.15.1.30>;tag=2e3c2071-c895-4069-afc2-37a19b20637a From: <sip:a-number@10.15.1.12;user=phone>;tag=SDkbo9901-42090 Call-ID: SDkbo9901-71a1d17456b829b4c422af61de9eee7e-ao32g50 CSeq: 1 ACK Contact: <sip:a-number@10.15.1.12:5060;transport=udp> *Route: sip:10.15.1.30;lr;ftag=SDkbo9901-42090 The problem: this ACK gets not retransmitted to Asterisk At first, I thought it was some sanity check but after disabling that I realized that it was in the WITHINDLG route. For the incoming ACK I get in the logs: Jun 8 11:56:47 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53240]: ALERT: <script>: Inside LOOSE route for ACK proto=UDP trans=4194304 from=sip:00754112601@10.15.1.12;user=phone route=sip:10.15.1.30;lr;ftag=SDkbo9901-42090 src_ip=10.15.1.12 And once the ACK is ready to be sent to Asterisk, the Route header has been removed and no Record-Route has been added so it fails. Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]: INFO: rr [rr_mod.c:402]: pv_get_route_uri_f(): No route header present. Jun 8 11:56:44 tone-0866-fe-2-qa /usr/local/sbin/kamailio[53238]: ALERT: <script>: ACK does not match transaction!! proto=UDP trans=4194304 from=sip:00754112601@10.15.1.12;user=phone route= src_ip=10.15.1.30 My WITHINDLG route looks like this: # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { xlog("L_ALERT","Inside LOOSE route\n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_ALERT","Inside LOOSE route for ACK proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n"); # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_ALERT","ACK does not match transaction!! proto=$rP trans=$mf from=$fu route=$route_uri src_ip=$si \n"); exit; } } sl_send_reply("404","Not here"); } exit; } } Thanks for reading this :) Any idea about how to validate the transaction? t_check_trans is not being validated... Cheers, Francisco.

2 2

rtpproxy question
by Albert Petit 09 Jun '16

09 Jun '16

Hi list I have installed a newer version of rtpproxy (2.0) in our development server , i was happy because it brings lot of performance improvements However i have found an issue when clients use TLS+SRTP In that scenario i do not want rtpproxy to decrypt/reencrypt the traffic as my B2BUA does that or in some scenarios we do plan to add end to end encryption. RTPProxy 1.X was doing that perfectly Instead since i use rtpproxy 2.0 all SRTP and SRTCP traffic is decrypted by rtpproxy and i receive it clear in the B2BUA How can i disable this new feature and that rtpproxy just keeps sending the received SRTP/SRTCP stream without decrypting it?

3 3

IPv6 Only client-rtp proxy configuration
by aman jeff 09 Jun '16

09 Jun '16

Hi team, How to do the Configuration of Kamailio+RTP-Proxy for SER in IPv6, Currently I can establish SIP Session,while rtp start call are not audible.So Some configuration need to change,Am bit confused on it,Please help to resolve the issue,For my proper VoIP-Communication in IPv6 Surveillance. Regards Arnab

3 2

kamailio mysql+ldap+radius authentication
by wanmon 08 Jun '16

08 Jun '16

Dear list, I have Kamailio 4.4.1 running with local mysql authentication (WITH_MYSQL/WITH_AUTH). i have RADIUS and LDAP services running. I would like to use all of them (MYSQL+RADIUS+LDAP) for Authentication. How is the best "kamailio.cfg" approach to use all defined users on RADIUS and LDAP? so, when a user try to login, kamailio will try MYSQL, them LDAP, them RADIUS. If user/password are not define on any of those 3 system, authentication will fail. Do i need modify my current LDAP tree? Im trying to using standart "userPassword" attribute. This is my reference: http://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth i would like to "add" route[LDAPAUTH], not to "replace" route [AUTH]. Kind Regards

2 1

A: README files for modules generated automatically
by Daniel-Constantin Mierla 08 Jun '16

08 Jun '16

Hello, following past discussions about different formats resulted when generating readme files for modules on various systems, which typically led to conflicts when backporting, I put in place a cron job running on kamailio.org to fetch from github the docbook files, generate the readme and push them back. The cron job is running every 15min. >From now on it is no longer needed to generate the readme yourself. Just commit the changes to xml files and push them to git repository. The readme will be refreshed automatically in a bit. The commits in this case are done by a generic git account named kamailio-dev. The commit message looks like: modules: readme files regenerated - app_mono ... Only the first module that has the readme updated is listed (in this example being app_mono). The readmes are automatically generated for master and 4.4 branches. If you do consistent changes to the docbook file, you can generate the readme locally to be sure there is no error, then restore the old version before committing - relevant commands: make modules-readme modules=modules/async rm modules/async/README git checkout modules/async/README Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

4.4 compiler warnings on debian jessie
by Juha Heinanen 08 Jun '16

08 Jun '16

I'll come back to mem/tlsf_malloc.c issue. Camille asked if the compiler warnings on mem/tlsf_malloc.c were occurring on x86 32 bit machine. I now tried and, yes, they do not appear on i386 jessie, but not on amd64 jessie. So looks like this is the same issue that Ovidiu has reported. -- Juha

2 3

4.4 fails to start on debian jessie due to tls issue
by Juha Heinanen 08 Jun '16

08 Jun '16

Was this fixed on master? Jun 8 08:12:46 sars /usr/bin/sip-proxy[19017]: : tls [tls_init.c:557]: init_tls_h(): ERROR: tls: init_tls_h: installed openssl library version is too different from the library the Kamailio tls module was compiled with: installed "OpenSSL 1.0.1t 3 May 2016" (0x1000114f), compiled "OpenSSL 1.0.1k 8 Jan 2015" (0x100010bf).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) Jun 8 08:12:46 sars /usr/bin/sip-proxy[19017]: CRITICAL: <core> [main.c:2514]: main(): could not initialize tls, exiting... If so, please backport to 4.4 since it is now impossible to start kamailio on debian jessie. -- Juha

2 6

ack with cscall-id header
by Rawad Khalil 07 Jun '16

07 Jun '16

Dear kamailio community, I have a problem that seems really weird. i have the same problem using kamailio 4.3 and 4.4.1. The scenario is the following: I make a call from a sip phone to kamailio. kamailio routes the Invite to an asterisk server. After a while asterisk sends a Bye request to kamailio. Now instead of routing the Bye to the sip phone, I want to create a re-invite request from kamailio and send it to the sip phone. The purpose of the re-invite is to connect the sip phone to another sip phone that already has an active call. This part is working fine but the problem occurs after the sip phone sends the ‘200 OK’ message for the re-invite. Kamailio sends an ACK for the 200 message but this ACK has a ‘CSCall-id’ header instead of the ‘Call-id’ header. To create the re-invite I am using the UAC module with one change to the source code. In the uac.c file found in the tm module folder, I added 2 lines of code to the request function in order to stop the UAC module from creating a from tag(attached). I also attached a trace that shows the ack message with cscall-id. Thank you, Rawad

2 1

Re: [SR-Users] Regarding kamailio-4.4.0 msrp module
by Daniel-Constantin Mierla 07 Jun '16

07 Jun '16

Hello, what is missing? What is highlighted seems to be in the code. Can you provide a patch or make a pull request on github project? That will make it simpler to understand what you want to change. Cheers, Daniel On 07/06/16 11:51, Surendra.P wrote: > > Hello Daniel, > > > > Switch case control is missing while parsing MSRP > pseudo variables. Please find the code snippet below: > > > > int pv_parse_msrp_name(pv_spec_t *sp, str *in) > > { > > if(sp==NULL || in==NULL || in->len<=0) > > return -1; > > > > switch(in->len) > > { > > case 3: > > if(strncmp(in->s, "buf", 3)==0) > > sp->pvp.pvn.u.isname.name.n = 1; > > else goto error; > > break; > > case 4: > > if(strncmp(in->s, "body", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 2; > > else if(strncmp(in->s, "code", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 3; > > else if(strncmp(in->s, "hdrs", 4)==0) > > sp->pvp.pvn.u.isname.name.n = 4; > > else goto error; > > break; > > case 5: > > if(strncmp(in->s, "msgid", 5)==0) > > sp->pvp.pvn.u.isname.name.n = 5; > > else if(strncmp(in->s, "conid", 5)==0) > > sp->pvp.pvn.u.isname.name.n = 21; > > else goto error; > > case 6: > > if(strncmp(in->s, "method", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 6; > > else if(strncmp(in->s, "buflen", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 7; > > else if(strncmp(in->s, "sessid", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 8; > > else if(strncmp(in->s, "reason", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 9; > > else if(strncmp(in->s, "crthop", 6)==0) > > sp->pvp.pvn.u.isname.name.n = 12; > > else goto error; > > break; > > > > > > Regards > > Surendra > > > > > > _______________________________________________ > 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 - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Regarding kamailio-4.4.0 msrp module
by Surendra.P 07 Jun '16

07 Jun '16

Hello Daniel, Switch case control is missing while parsing MSRP pseudo variables. Please find the code snippet below: int pv_parse_msrp_name(pv_spec_t *sp, str *in) { if(sp==NULL || in==NULL || in->len<=0) return -1; switch(in->len) { case 3: if(strncmp(in->s, "buf", 3)==0) sp->pvp.pvn.u.isname.name.n = 1; else goto error; break; case 4: if(strncmp(in->s, "body", 4)==0) sp->pvp.pvn.u.isname.name.n = 2; else if(strncmp(in->s, "code", 4)==0) sp->pvp.pvn.u.isname.name.n = 3; else if(strncmp(in->s, "hdrs", 4)==0) sp->pvp.pvn.u.isname.name.n = 4; else goto error; break; case 5: if(strncmp(in->s, "msgid", 5)==0) sp->pvp.pvn.u.isname.name.n = 5; else if(strncmp(in->s, "conid", 5)==0) sp->pvp.pvn.u.isname.name.n = 21; else goto error; case 6: if(strncmp(in->s, "method", 6)==0) sp->pvp.pvn.u.isname.name.n = 6; else if(strncmp(in->s, "buflen", 6)==0) sp->pvp.pvn.u.isname.name.n = 7; else if(strncmp(in->s, "sessid", 6)==0) sp->pvp.pvn.u.isname.name.n = 8; else if(strncmp(in->s, "reason", 6)==0) sp->pvp.pvn.u.isname.name.n = 9; else if(strncmp(in->s, "crthop", 6)==0) sp->pvp.pvn.u.isname.name.n = 12; else goto error; break; Regards Surendra

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 07 Jun '16

07 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Admin Arnab

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 07 Jun '16

07 Jun '16

Hi Team, How to configure rtp proxy for IPv6 Only client,now we are not able to get the audio, Please help Regards Arnab

2 1

Kamailio Sorce code Documents
by aman jeff 07 Jun '16

07 Jun '16

Hi team, I like to know the source code documention of kamailio, I like to customize textops.c,like to get the variables and structure documents gparam_p, etc.. Regards Jeff

2 1

Yealink BLF
by Bill 06 Jun '16

06 Jun '16

Has anyone been successful in getting Yealink BLF's to work with Kamailio presence It appears that the xml format being sent is not compatible with Yealink, butr I can't find how to change the format of the body. Bill

2 1

RTPENGINE not kernalizing in some cases
by Dmitry 06 Jun '16

06 Jun '16

is it possible that RTPENGINE failes because 2 rtpengine_offer (2 invites during autharization) and the 183 Reply as I see in the log - is seen as an ANSWER Both INVITES are with SDP RTPENGINE_MANAGEThis call is not answered Received command 'offer' from 127.0.0.1:54490 Creating new call] offer time = 0.000134 sec Replying to 'offer' from 127.0.0.1:54490 Received command 'offer' from 127.0.0.1:53879 offer time = 0.000060 secReplying to 'offer' from 127.0.0.1:53879 Received command 'answer' from 127.0.0.1:40099 answer time = 0.000169 sec Replying to 'anser' from 127.0.0.1:40099 ICE negotiated: local interface 10.34.101.104Confirmed peer address as 10.34.101.104:7458Confirmed peer address as 10.34.101.103:62608

1 0

IPv6 Only client-rtp proxy configuration
by aman jeff 06 Jun '16

06 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Arnab

1 1

Wrong Publish status
by Aleš Šturm 06 Jun '16

06 Jun '16

Hello, Kamailio 4.3: I am testing Presence module with more complex scenario. I have 2 SIP clients (presentity) with equal identities (3918) and one watcher (2025). All my clients are Monster 0.9.25. My Kamailio Presence module is configured with "retrieve_order=1" (presentity records are retrieve by priority order). I have situation when wrong status is send to the watcher (2025). 1. First all three clients registers and publish its own status. Below is print from my MySQL database (presentity table) mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340687 | 1464337087 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 2. Then first 3918 client publishes his mood >11111<. The status in >Presentity< table is refreshed under the id=84480. NOTIFY is sent to 2025 with status with mood >11111<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 3. After that the second 3918 client publishes his mood >22222<. The status in >Presentity< table is refreshed under the right id=84481. NOTIFY is sent to 2025 with status with mood >22222<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 4. Then finally, the first 3918 changes his mood to "33333". The status in >Presentity< table is refreshed under the right id=84480. BUT, to 2025 is sent NOTIFY with mood "22222". "received_time" timestamp of status with mood "33333" is higher than timestamp of status with mood "22222". BUT the "id" attribute of "presentity" table of status with mood "33333" is smaller. I think that statuses with equal "priority" are retrieved by "id" and not by "received_time"!? mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340729 | 1464337129 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>33333</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | The Wireshark trace is attached. Can anyone help. Please? We have a big installation in the Gazprom's (Rusia) network, where Kamailio is used as a Presence server only. Thanks Ales

2 8

Yealink BLF
by Bill 05 Jun '16

05 Jun '16

Has anyone been successful in getting Yealink BLF's to work with Kamailio presence It appears that the xml format being sent is not compatible with Yealink, butr I can't find how to change the format of the body. Bill

1 0

rtpproxy ipv6 problem
by aman jeff 03 Jun '16

03 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Arnab

1 0

Kamailio 4.4.1 REGISTER Parse Error in PCSCF
by Mihail Dakov 02 Jun '16

02 Jun '16

Hi Folks, I have the strangest problem just after installing latest stable version. I had tried both ways using the repo (http://deb.kamailio.org/kamailio44) and downloading code with local compile (https://www.kamailio.org/w/2016/05/kamailio-v4-4-1-released/) The initial scenario is to get M2M with commercial IP and subsequently use the Rx interface to do the same. We are blocked because we cannot even achieve clean register for the UEs. UE1(galaxy s4 zoiper) -------- Kamailio IMS aio | UE2(galaxy s4 zoiper) -------- | What happens is that the first register message triggers some quasi-endless loop in the PCSCF. Hence it cannot be forwarded to SCSCF, ICSCF and HSS. But this doesn't always happen and between all the parse problems there are some register messages which get passed to the scscf icscf and hss. Somehow the parser feeds itself from the previous parse run and thus loops quite intensively. The message I get is SIP/2.0 400 CSeq method does not match request method. I must say that reiterating the registration process sometimes i get the register and sometimes i don't. Has anyone encounter similar issue for similar use case and can share some thoughs about a fix? I attached relevant log file, pcaps and configuration files. Let me know if more config files are needed.

3 15

Nonce extract from SIP Meesage
by Sergio Serrano 02 Jun '16

02 Jun '16

Hi all, I just try integrate Kamailio with external software and We want taht kamailio doesn't authenticate users. I want that the autehtication process would be done by external software. I want To receive a REGISTER for a user. Then I send a 407 and then when I receive all paramters for autehtincation(nonce, response, digestURI, etc, etc etc) send that data to external software and receive if data is correct, then sent 200 OK and save location in external software. Bu t I can see any pseudovariable or enything else to obtain that dato to provide to external software. Anyone could think that this process could be done? Regards, Sergio

3 2

NSQ libevbuffsock dependency
by smititelu 02 Jun '16

02 Jun '16

Hi, Upon trying to compile latest Kamailio from sources I got an: " nsq.h:9:24: fatal error: evbuffsock.h: No such file or directory #include <evbuffsock.h> ^ " Looking on the new NSQ module documentation I see the libevbuffsock dependency. Is there other recommended way, to install the library besides compiling from sources at [1]? Thx, Stefan [1] https://github.com/mreiferson/libevbuffsock

2 2

sip trace
by Jeremy Betts 01 Jun '16

01 Jun '16

I have a new carrier that wants sip traces of certain scenarios. They also require IPSEC tunnelling for all SIP messages. So when I use tcpdump or any other similar tool I can see the incoming SIP messages but outgoing SIP messages are ESP encoded. I've played with the siptace module but can't seem to get it to log replies. Does anyone have any solutions?

2 2

Dynamic advertised ip for rtpengine/rtpproxy
by Serge Yuriev 01 Jun '16

01 Jun '16

Hi, We have several exits from our network and all of them natted. Could we set adv ip for rtpproxy/rtpengine from db in realtime or we must define them on daemon run time only? -- Wbr, Serge via mobile -- Serge via mobile

4 3

Send Register from Kamailio
by Grant Bagdasarian 01 Jun '16

01 Jun '16

Hello, Which module do I need to have Kamailio send REGISTER messages to our carrier? Is it the UAC module or REGISTRAR module? It just needs to register with our carrier in order to receive inbound calls. Regards, Grant

2 4

advertise xxx.xxx.xxx.xxx:5060 is weird
by Sébastien Brice 01 Jun '16

01 Jun '16

Hi guys, I have a local proxy running on private ip 192.168.3.97 and advertising the wan ip of my box (THEPUBLICIPOFMYBOX) this local Kamailio is a stateful proxy that TM REGISTERS and INVITES to a Kamailio registrar(PUBLICIPKAMAILIOREGISTRAR) (please have a look at the kamailio.cfg attached to this email.) I want to force the signalling and sessions through this local proxy (some kind of outbound proxy) I have jitsi and xlite behind (ip 192.168.3.39 and 192.168.3.16) that tries to call themselves. user agents keep retransmitting 200 OK because they never get the ACK following the 200 OK pickup. If i remove the advertise THEPUBLICIPOFMYBOX of the kamailio.cfg the three handshake is working ok and the call is no longer dropped after a few second! i have a true bidirectionnal session and that's working ok. The bad thing is RTP flows straight through router (thanks to rtpproxy i guess) but i want incoming and outgoing RTP traffic to go on central local node 192.168.3.97 How can i achieve that? thanks. PS: i am not using path header because i am not skilled enough and kamailio is really hard horse to train

2 5

DHCP renewal conflicts with kamailio?
by Joonas Harjumäki 01 Jun '16

01 Jun '16

Hi! The attachment contains a log trace of kamailio 3.3.2. There is some errors in the log and I wonder if it is normal operation? The errors appeared in the middle of DHCP lease renewal and I wonder if the configuration (of dhclient/kamailio) could be improved to prevent such errors? Or to make sure kamailio retries udp_send sufficiently many times. The kamailio operated normally before renewal procedure began and after the procedure finished. Joonas Harjumäki

2 1

how to set kamctl to find ctl anf fifo that are not at /var/run
by Uri Shacked 01 Jun '16

01 Jun '16

Hi, I am using centos 7. By default on centos 7, the var/run file system is tmpfs. So, when kamailio runs with user and group other than root, it can not create the fifo file. I set the kamailio service and config files to work on usr/local/var/run/kamailio. Now, how do i set the kamctl to work with the fifo on /usr/local/var/run/kamailio ? Uri

3 3

Want dialog.default_timeout NOT to reset on subseq. req
by Andrey Utkin 01 Jun '16

01 Jun '16

Hi! I'd like to restrict max call duration. Currently we have something that partially works: dialog.default_timeout. But, as described in "How it works" section http://www.kamailio.org/docs/modules/devel/modules/dialog.html#idp2029360 "The dialog timeout is reset each time a sequential request is processed." So this option works for us when the useragents don't send sequential UPDATEs or reINVITEs. I am not sure if it is adequate to ask for this feature in dialog module. I am aware of possibility to terminate any call by management command, and of cnxcc module which has call duration control. I would love to find something that simple - a single option for max call duration, without external mechanism (cronjob etc.) or full-blown billing. Any suggestions are very welcome. Thanks in advance. Please reply to all recipients, I am not subscribed to this maillist.

2 1

513 Message too big for VIDEO calls
by Varghese Paul 31 May '16

31 May '16

Hi all, I am trying to setup WSS on opensips-2.2. I am getting "513 Message too big " when i am trying to make video calls. Audio calls and Registration is working fine.I tried to change the message_chunk_size in WSS module but still i am getting the message. Client : SIPML5 and JSSIP Opensip version : 2.2 I am attaching the sipml5 console logs. Please let me know is there any settings i can apply to resole this. Thanks in Advance Varghese Paul

2 1

Extending has_body() with additional mime types
by Giacomo Vacca 31 May '16

31 May '16

Hi all, apologies if this is an old topic but I've noticed that has_body() can provide false positives and would like to discuss a solution. As a practical example, has_body("application/vq-rtcpxr") was returning 1 for Content-Type "application/dialog-info+xml". The interpretation is that they are both unknown and so they match. Using 'if ($cT == "application/vq-rtcpxr)" provides a work around, but I wonder what's the suggested solution for this? a. Add the missing mime types (from RFC 6035 and 4235 respectively) to the list b. Instead of returning 1, return another code, e.g. 2, when the mime type is unknown c. Switch to string comparison when the mime type is unknown d. Something else? Thanks, Giacomo

2 1

Accept INVITE with transport=tcp
by Nahum Nir 31 May '16

31 May '16

Hi, I have a client that sends INVITE with transport=tcp. How can I make Kamailio to work with that? Thanks and regards, Nahum

2 1

Issue with LCR an failover routers
by Alberto Sagredo 30 May '16

30 May '16

Hi Im using LCR module and trying to handle 503/408 errors on provider to route to a second one. Actually im using if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Gateways no disponibles"); sl_send_reply("500", "Server Internal Error - No gateway"); exit; } else { xlog("GW Seleccionado '$avp(i:709)'\n"); xlog("Dominio de destino: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("No hay GW de Backup"); sl_send_reply("503", "Service not available - No gateway"); exit; } else { xlog("Enviado por primer GW\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); t_on_failure("FAILBACK"); } failure_route[FAILBACK] { if (t_is_canceled()) { exit; } if (t_check_status("408|50[34]")) { route(LCR_OUT_OP2); } } And using for OP2 same routing block with id=3... This second providers use AUTH so using: t_on_failure("TRUNKAUTH"); failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { if (to_gw("3")) { xlog("Autenticación "); $avp(auser) = "user"; $avp(apass) = "pass"; } uac_auth(); t_relay(); exit; } But getting this error: May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [t_fwd.c:1704]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [tm.c:1466]: _w_t_relay_to(): ERROR: w_t_relay_to: t_relay_to failed Any idea how to resolve that?

2 2

converting integer to string
by Henry Fernandes 30 May '16

30 May '16

In Kamailio 4.2, I’m having problems converting an AVP integer value into a string. My Kamailio config stores the return code from auth_check() and outputs it. $avp(auth_code) = auth_check("$fd", "subscriber", "1"); xlog("L_NOTICE", “response = $avp(auth_code)”); When I check the AVP against an integer, it behaves normally. However, I cannot convert it to a string. The return value -3 (invalid user) gets converted to the string “18446744073709551613” when outputting it. My hunch is that Kamailio is treating the AVP as an unsigned integer (rather than a signed integer) when it converts it to a string. If I try this with an AVP with a positive integer, it works as expected. Can someone tell me how to convert a negative integer into a string in Kamailio? -H

3 3

Variable in dialog event routes
by Minh Phan 30 May '16

30 May '16

Hi List, I'm trying to have an external application to write CDR to a database by doing this: event_route[dialog:start] { evapi_relay("{'cdr_id':* $dlg_var(cdr_id),* 'event': 'answered'}"); } event_route[dialog:end] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'ended'}"); } event_route[dialog:failed] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'failed'}"); } $dlg_var(cdr_id) is set inside a request route. However, in these event routes, its value is always null. Its seems that $dlg_var variables are not available in those event routes. What are the variables available in dialog event route? Any suggestion on how to achieve what I'm trying to do? Best regards, Minh

2 3

http_async_client segfault
by Jason Penton 30 May '16

30 May '16

Hi, Has anyone experienced segfault using http_async_query from http_async_client module? Here is BT: #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 53 iofgets_u.c: No such file or directory. (gdb) bt #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 #1 0x00007fca68f08afa in get_contents (stream=<optimized out>, len=1032, linebuf=0x7fca23ffe660 "127.0.0.1") at nss_files/files-XXX.c:201 #2 internal_getent (result=0x7fca23ffe5c0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, af=af@entry=0, flags=flags@entry=0) at nss_files/files-XXX.c:246 #3 0x00007fca68f09ca3 in _nss_files_gethostbyname4_r (name=name@entry=0x7fca2e790fa0 "localhost", pat=pat@entry=0x7fca23ffebe0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, ttlp=ttlp@entry=0x0) at nss_files/files-hosts.c:402 #4 0x00007fca79eb7757 in gaih_inet (name=<optimized out>, name@entry=0x7fca2e790fa0 "localhost", service=<optimized out>, req=req@entry=0x7fca341c7490, pai=pai@entry=0x7fca23ffed10, naddrs=naddrs@entry=0x7fca23ffed00) at ../sysdeps/posix/getaddrinfo.c:850 #5 0x00007fca79ebae0d in __GI_getaddrinfo (name=0x7fca2e790fa0 "localhost", service=0x7fca23ffeec0 "18000", hints=0x7fca341c7490, pai=0x7fca23ffee78) at ../sysdeps/posix/getaddrinfo.c:2406 #6 0x00007fca6e0f80f4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #7 0x00007fca6e104fa4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #8 0x00007fca6e10284b in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #9 0x00007fca723c9182 in start_thread (arg=0x7fca23fff700) at pthread_create.c:312 #10 0x00007fca79ee547d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 Camille, are you maintaining this module now?

2 2

Multiple async operations in single transaction fail (using http_async_client)
by Thom Seddon 30 May '16

30 May '16

Hi All, I've been doing some work this week with the new http_async_client module, it seems that with it's default behaviour it's not possible to execute another async query in the block immediately following the response, for example the following config: request_route { if (!t_newtran()) { sl_reply_error(); exit(); } http_async_query("http://localhost:8090/req_1", '{"user":"$fU"}', "REQ_1"); } route[REQ_1] { xlog("L_INFO", "route[REQ_1]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_1]: status $http_rs\n"); xlog("L_INFO", "route[REQ_1]: Sending REQ_2\n"); http_async_query("http://localhost:8090/req_2", '{"user":"$fU"}', "REQ_2"); } else { xlog("L_INFO", "route[REQ_1]: error $http_err)\n"); } } route[REQ_2] { xlog("L_INFO", "route[REQ_2]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_2]: status $http_rs\n"); } else { xlog("L_INFO", "route[REQ_2]: error $http_err)\n"); } } Generates the following error: 10(11832) INFO: <script>: route[REQ_1]: INIT 10(11832) INFO: <script>: route[REQ_1]: status 200 10(11832) INFO: <script>: route[REQ_1]: Sending REQ_2 10(11832) WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [53195:560306836] Looking at the source, http_async_client calls t_suspend, then goes away and performs the request in the background. On response it calls t_continue and then this block <https://github.com/kamailio/kamailio/blob/4.4/modules/tm/t_suspend.c#L270-L…> in t_continue will continue execution, the second query will cause http_async_client to call t_suspend whilst sill executing from the previous t_continue. Once they query is sent, the first t_continue will complete, removing the T_ASYNC_SUSPENDED flag. When the second query returns it doesn't find the flag and so fails as shown. I'm far from an expect on this so would really appreciate some pointers, it seems to fundamental to be a race condition in tm but the usage in http_async_client seems to match usage in other modules so I'm not sure on the best tactic. I've tried peppering append_branch and async_route call's in various places to no avail. Full debug log here: http://pastebin.com/raw/rFmm0fmf Any help appreciated, *Thom Seddon*

2 1

username of Contact header
by Al S 27 May '16

27 May '16

Hi, I am trying to read username portion of Contact header: My Contact header content has a URI value such as: sip:8131110000@10.10.10.10.:5060 and I am trying to read : 8131110000 I tried the following perl similar RE and it didn't work: ($var(main_number2)) = $ct =~ /sip:(.*)@/; Thanks,AS

2 1

4.4: Kamailio crashes at start - sometimes
by Sebastian Damm 27 May '16

27 May '16

Hi, we have a machine running 16 Kamailio instances, and while upgrading to 4.4.1 (from 4.3.5), 8 of them wouldn't start. When downgrading to 4.3.5, they all start again. All of them have pretty identical configuration files, except IPs, ports and some code enabled or disabled via defines. After comparing a working and non-working configuration and adjusting setting by setting, we finally ended up with a working configuration. The difference is, that it won't start when a part of the code DOES NOT get included. If it gets included, it will start. This is the mentioned part of our main route: #!ifdef ENABLE_INV_RATELIMIT # Check for INVITE limit if (is_method("INVITE") && $au == $null && !($ua =~ "sipgate") ) { $var(invcount) = $shtcn(invcount=>%~$fU); xlog("L_INFO", "INVITE Requests from $fU in last 30 seconds: $var(invcount)\n"); if ($var(invcount) < 12) { $var(uniqcid) = $ci + $Ts + $ft; $var(tkey) = $fU + '-' + $(var(uniqcid){s.md5}{s.substr,0,10}); $sht(invcount=>$(var(tkey))) = 1; $var(uniqcid) = $null; $var(tkey) = $null; } if ($var(invcount) > 10) { if ($var(invcount) == 11 ) { xlog("L_NOTICE", "User $fU ($var(domain2use)) over ratelimit for new calls, rejecting.\n"); } # Enable this only after evaluating the impact! append_to_reply("Retry-After: 30\r\n"); sl_send_reply("503", "Call Rate Limit Exceeded"); exit; } } #!endif If we put this line at the top of the configuration file, everything works: #!define ENABLE_INV_RATELIMIT If we delete this line, startup does not work. It just sits in ps for one minute without forking, and then gets terminated. We enabled a bit of debugging, and this is apparently the error causing Kamailio to shutdown: May 25 14:50:15 kammel /usr/sbin/kamailio[24989]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 53: nathelper May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0x9f0428 and 0xa34428 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0x9f0240, 82, 1, (nil)), fd_no=0 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Bad file descriptor [9] May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1747]: tcp_receive_loop(): failed to add tcp main socket to the fd list May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1815]: tcp_receive_loop(): exiting... I have no idea, how this part of the code could lead to this error, but it is reproducable, that at least on this system setting or disabling this define fixes or breaks the startup. Does anybody have an idea, what's happening there? Best Regards, Sebastian

2 4

Multihomed kamailio - one IP for each account
by Attila Megyeri 27 May '16

27 May '16

Hi Folks, I am looking for a solution to do the following, basically an SBC-like solution. I have an internal asterisk system that should make outbound calls towards a provider, but that provider authenticates the customer based on IP address. And I have many such customers. The provider is not willing to make changes to his practice... Asterisk is bound to a single IP, so there is not much I can do. I was thinking about creating several outbound accounts for asterisk, and then a multi-homed Kamailio would do the following: If request comes from user1, Kamailio would send the request to the provider from IP1. If request comes from user2, Kamailio would send the request to the provider from IP2. And so on, for as many IPs as many customers I want to handle. Is there a more elegant way to do this? If not, how should I implement this? Thanks in advance, Attila

2 4

Kamailio Via: header issue
by Laura 27 May '16

27 May '16

Good morning list, I need to ask you a question about a problem we are fighting on our VoIP platform based on Kamailio 4.3.5. Our platform is made by more nodes over Europe countries, and we are encountering a problem with the size of the SIP package which, exceeds the physiological limit of about 1350 bytes. The real problem is caused from Via: Header (inside the SIP packet) which is added to each element of the system for transit on the call. >From what we have analyzed, the problem is not so much due to the number of entered Via: records but, by the fact that they contain a branch parameter extremely long. Here the call flow example from CARRIER A —> Kamailio1 —> Kamailio2 —> CARRIER B Extract from SIP message sent from CARRIER A to K1 INVITE sip:999912341234@192.168.158.42 SIP/2.0. Via: SIP/2.0/UDP 192.168.220.141:5060;branch=z9hG4bK49e4301c;rport. Extract from SIP message sent from K1 to K2 INVITE sip:999912341234@192.168.127.244 SIP/2.0. Record-Route: <sip:192.168.158.42;lr;did=194.a0f1;nat=yes>. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bK0635.7c1e97c2d5c01ea98e8d0e7fa23a3822.0. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.cuOBV4OBVLMGZfWxF-W.y6Mx1LgRWIC9gIgx4fzxj7MBP7MBVAOBF4M.1jmxuqC93X3heroEWvH9vsCFN43qd4zRj4Mlyf3l1LNSQLpx4uMx3A. Extract from SIP message sent from K2 to CARRIER B INVITE sip:999912341234@192.168.65.248 SIP/2.0. Record-Route: <sip:192.168.127.244;lr;did=194.5122;nat=yes>. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bK0635.f843472880b380acde3a33732975fde9.0. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1LNSQLpx4uMx3AzuaVHRaYpB1JNEt736cQkBIvalaJmly6Mlj7W6Mfg.qw3leqWRMAMRKrz.rIzSPAg.pE3.Vl3.MZMBV7My**. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1JNEt736cQkBIvalaJmwWLORv4mK0Hot3w.jpam6t4kRWWOEWu.4eRWFQGKqfWauYEKwrSOKN7k.gWkxtMgue9mjMsg4IhkxaEkhrbW4uGjxpWPIg0.4eSWF47MRY1kUKfPlqlRxWvg9e5aKp6mxI6N4WS.BPlNRP4kIebWFudkR3loxtM.IWa.U0ZWUKWkxW0. Is there anyone here with the idea how to solve this problem ? Best regards

3 8

rtpproxy module question
by gmele 27 May '16

27 May '16

Hello, we are using an RTP Proxy from rtpproxy.org as media relay to establish communication between our mobile phones. Of course, we are using the kamailio rtpproxy module to modify the SDP payload and control the proxy. In our Kamailio configuration, we have 1 kamailio configured as Proxy and one kamailio configured as Registrar. So calls go through the Proxy and then to the Registrar who will update the SDP header and select an available rtp proxy. We have noticed that sometimes, the rtp udp flow between the phones isn't routed properly by the rtpproxies, ending in the communication drop (all the SIP nego is working well, and the SDP are correctly patched with the rtp proxy address and port). Analyzing the RTP proxy packets, we have found that the Kamailio registrar gives the Kamailio proxy ip address in the RTP proxy create session command, but keeps the original sdp port. command looks like this: Uc96,101 DC -PbO~Wnm <proxy_ip> <port from original sdp phone packet> PZU5OITCW;1 We are using rtpproxy_manage() without any flags. It seems to us that this ip and port are used as default forward route as long as the callee hasnt connected to the rtpproxy. *Is it correct? * If its true, Its seems to us that this cant work as we are mixing the proxy sip address with a udp port open on the phone ? *Is our analysis correct ?* Can we use some option in rtpproxy_manage to replace the proxy ip by the phone ip as seen in the via route ? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/rtpproxy-module-question-tp148850.h… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio not sending reply
by Jeremy Betts 26 May '16

26 May '16

Hello, I'm having an issue where kamailio is not sending the failure reply as I would expect when receiving calls from a certain provider. Kamailio sends the 403 response on calls from other providers. The big difference I see is that the "problem" provider is using compact headers. I am trying to force the 403 response to be sent for testing purposes but I just can't get it to reply to this one provider. I've included traces and configuration excerpts below. Any help is much appreciated! Problem Call (No Reply Sent): U 63.79.178.192:5060 -> 184.171.164.100:5060 INVITE sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. Record-Route: <sip:63.79.178.192;lr>. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 INVITE. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144. Max-Forwards: 27. m: <sip:199.173.94.144:5060;transport=UDP>. k: 100rel, resource-priority, replaces. c: application/sdp. l: 235. P-Asserted-Identity: <sip:+17143258018@63.79.178.192;user=phone>. Privacy: none. . v=0. o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106. s=-. p=+1 6135555555. c=IN IP4 199.173.68.106. t=0 0. m=audio 55380 RTP/AVP 18 0 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=fmtp:18 annexb=no. # U 63.79.178.192:5060 -> 184.171.164.100:5060 ACK sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 ACK. l: 0. Max-Forwards: 27. . Successful Call (Reply Sent): # U 64.136.173.31:5060 -> 184.171.164.100:5066 INVITE sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. Record-Route: <sip:sansay3408006464rdb14345@64.136.173.31:5060;lr;transport=udp>. To: <sip:7146466334@184.171.164.100>. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Contact: <sip:17143258018@64.136.173.31:5060>. Supported: timer. Session-Expires: 1800;refresher=uac. Min-SE: 90. P-Asserted-Identity: <sip:17143258018@192.168.20.76>. Privacy: none. Expires: 120. Max-Forwards: 67. Content-Type: application/sdp. Content-Length: 274. . v=0. o=Sansay-VSXi 188 1 IN IP4 64.136.173.31. s=Session Controller. c=IN IP4 69.85.185.142. t=0 0. m=audio 37970 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 184.171.164.100:5066 -> 64.136.173.31:5060 SIP/2.0 403 DID Lookup Failed. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Server: x-Freevoice SIP Proxy 4.21. Content-Length: 0. . # U 64.136.173.31:5060 -> 184.171.164.100:5066 ACK sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 ACK. Max-Forwards: 70. Content-Length: 0. Kamailio Config: request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { setflag(FLT_ACCFAILED); if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(is_method("INVITE")||is_method("MESSAGE")){ route(ORIGINATE); #route(FROMAST); } route[ORIGINATE] { sl_send_reply("403", "DID Lookup Failed"); exit; } Jeremy Betts <https://www.youtube.com/user/FreevoiceLLC> <https://www.youtube.com/user/FreevoiceLLC>

2 2

webrtc tlsv1 alert access denied
by Arsen 26 May '16

26 May '16

Hi guys! I am trying to configure kamailio with WSS. We have trusted certificate installed SIP over TCP/TLS works fine. But when I try WSS I got error: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL routines:SSL3_READ_BYTES:tlsv1 alert access denied ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading Before above error it was 'bad certificate', so I have imported CA in the firefox and now I get these errors.. I have tried sipml5 and tryit.jssip.net same issue with both clients, also it seems I have these errors only when I use firefox, when I use chrome it even doesn't show me an error.. Any ideas? Thanks! -- Regards, Arsen.

2 3

SRV and dialog stickiness
by Alex Balashov 26 May '16

26 May '16

Hello, Sorry if this is a tired, worn question, but I've not dealt much with Kamailio's SRV support before: If a registrant has a contact binding whose domain component is subject to an SRV lookup with load-balanced or weighted entries, how does one solve the problem of ensuring that subsequent in-dialog requests go to the same host as the initial INVITE? Does Kamailio offer some facility for doing this? Is it somehow accommodated by SIP? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

rtpengine and SAVPF
by Ján Füri 26 May '16

26 May '16

Hi all, Im struggling with incoming calls to webrtc clients. I am trying to change INVITE's SDP using rtpengine from RTP/AVP or RTP/SAVP to RTP/SAVPF towards my webrtc clients. This works well. But if I send INVITE with SDP that includes RTP/AVP and RTP/SAVP as well, rtpengine creates incorrect SDP. It includes twice m=audio RTP/SAVPF and my WebRTC client does not like this INVITE and forbiddeds it by 488 Not Acceptable Here. Example of my incorrect INVITE : Content-Type: application/sdp Content-Disposition: session Content-Length: 1393 v=0 o=SBC 1464238673 1464238674 IN IP4 <MyPublicIP> s=SBC c=IN IP4 <MyPublicIP> t=0 0 m=audio 30454 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30455 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5X2wgU83bIwrwkHSymjnw48SiMGa1n+eIl50yr99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:iglHZqry a=ice-pwd:eyVXbFNoviYYlu5uuRZlnixFwQ a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30454 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30455 typ host m=audio 30484 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30485 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ozDJ3G/wmaYedQbcTafbhaTIt6raIJa6ugrLUC99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:gM1viWqu a=ice-pwd:zdS4tP1Mj4hLTe6huBWYrhtI1s a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30484 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30485 typ host My setup : incoming call -> MediaServer (freeswitch) -> SIP & Websocket proxy (Kamailio + Rtpengine) -> webRTC clients Is there anybody who could help me with this issue or explain me how to remove the second line of m=audio RTP/SAVPF ? -- Jan

1 0

Sequential calls messes up presence with pua-dialog info
by Ali Pey 25 May '16

25 May '16

Hello, We are using pua-dialoginfo and presence modules for BLF. When the first call comes in or goes out, every thing works great; however, a second call to the same key/extension (e.g a call waiting which is not answered) would turn off the key even though the original call/dialog is still active. Has anyone had experienced such problem? How can I fix this? Any constructive suggestion would be appreciated. Regards, Ali Pey

4 5

nh_reload_rtpp not available
by david 25 May '16

25 May '16

dear all i'm using kamailio 4.4.1 and i'm not seeing nh_reload_rtpp as a fifo exported function i'm trying to setup the dburl for the rtpengine sets using db_text but when i do kamctl fifo which i see all the commands of the module except the reload_rtpp nh_enable_rtpp nh_show_rtpp nh_ping_rtpp nh_show_hash_total /usr/local/kamailio/sbin/kamctl fifo nh_reload_rtpp 500 command 'nh_reload_rtpp' not available these are the config lines i have for that #!define DBURLRTP "text:///var/log/rtpengine_conf" loadmodule "db_text.so" modparam("rtpengine", "db_url", DBURLRTP) modparam("rtpengine", "table_name", "rtpproxy") and the database is like root@test:/var/log/rtpengine_conf# ls rtpproxy version root@test:/var/log/rtpengine_conf# cat version table_name(str) table_version(int) rtpproxy:1 root@test:/var/log/rtpengine_conf# cat rtpproxy set_id(int) url(str) weight(int) disable(int) 1:udp\:x.x.x.x\:x:100:0 2:udp\:x.x.x.x\:x:100:0 is there something i missed here? do i must use mysql or postgres drivers for this? (i use a test machine to old to install a driver of those with kam 4.4 maybe?) best regards david

1 1

doubt using rate limit
by david escartin 25 May '16

25 May '16

Hello Daniel i had the call rate for some minutes actually. maybe 4-5min and all the time the behaviour was like that. it's true that in that kamailio there wa no any other traffic, only those test calls i set with a sipp to simulate the load best regards david Hello, for how long you got this 20req/second? The module is not counting like continuous time, but more like traffic on each second, iirc. Cheers, Daniel On 23/05/16 11:24, david escartin wrote: > Hello all > > i have a quick question about using module ratelimit on kamailio 4.2 > or 4.4. > on tests made before running it on production, i set the RED > algorithm with a limit of 10 calls per second, but i saw the call > limitation was not done until we got more than 20 calls per second or > so on the queue. we didnt get any INVITE discarded. > after surpassing around 20 INV/sec, it's true the algorithm starts to > work fine and only 10INV/sec are passed more or less > > on production i have no easy way to check this behaviour, besides it's > a feature barely used. > > so i only want to know if this behaviour is known, or if i'm missing > soemthing > > my config is like this > > modparam("ratelimit", "timer_interval", 5) > modparam("ratelimit", "queue", "4:INVITE") > modparam("ratelimit", "pipe", "4:RED:10") > > route[RATE_LIMIT] { > $var(limitation)="4"; > if (!rl_check("$var(limitation)")) { > send_reply("505","Limiting"); > xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, > rejecting with 505\n"); > route(CLEAR); > exit; > } > } > > > thanks a lot and regards > david escartin > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio for sip routing architecture
by Loic Chabert 25 May '16

25 May '16

Hello Kamailio users, I'm working on a sip infrastructure, with opensource technologies, and of course Kamailio will run on my new infrastructure :) Kamailio will be my central routing engine, between SBC (running on asterisk), and somes boxes (asterisk too) where customers will be registred. Please find below my future infrastructure: ---------- -------------- -------- ---------------- | SBC1 | - - - - - | - - - -| kamailio1 |- - - - - | - - - - - | CB1 |-----------------| SIPPHONE | ---------- | -------------- | -------- ---------------- | | ---------- | | -------- | SBC2 | - - - - - | | - - - - - | CB2 | ---------- | | -------- | -------------- | ---------- | - - - -| kamailio2 |- - - - -| -------- | SBCn |- - - - - | -------------- | - - - - - | CBn | ---------- | | -------- Kamailio high availabality will be done with loopback address and BGP in two datacenters. SIP carriers will be connected to SBC and customer to one CB. I have made some tests with kamailio, and i use LCR to route call from CB to SBC, using wheigt and prorities to send call to carriers. Then, when a call arrived from carriers, i use DNS NATPER lookup to find the right CB box. CB HA is done using DNS SRV. So user is registred at one location, and only one. No location sharing beetwen boxes. I have some question concerning my architecture, and technologies used to routing call. Is LCR is the best option for routing call between multiple carriers ? DNS NAPTER is a modern solution to find an endpoint on this architecture ? Existing a better way to route call with kamailio, for exemple rt-json and curl ? In my point of view, kamailio should be a routing engine, using internal technologies, to maximise performances. Less work with kamailio and external app, best performances in sip routing. Is it the good vision ? Thanks in advance.

2 4

Kamailio routing to different carriers based on prefix
by José Seabra 25 May '16

25 May '16

Hi there, I'm looking for a solution to route calls based on prefix, my routing table might reach a long list of prefixs. I looked to carrierroute and it seems to be nice for this purpose but i have some doubts to how to handle correctly with carrierroute. IfI understood correctly, if I set the parameter avoid_failed_destinations to 1, can i use the same carrier id and domain on failure route to select another destination that matches the prefix? Can I have the same carrier ID with diferent domains sharing the same prefixs but mapping it to different destinations? Thank you for your help. Best regards -- José Seabra

2 1

call count / and work with despatcher
by jaflong jaflong 24 May '16

24 May '16

Hi Everyone, What is a accurate way to get current call count. I am only relaying calls. Initialy I tried the code below. It seemed to work, it reported the correct number of calls I had dialed. I get the result through kamcmd htable get command. ------------------------------------------ under route[RELAY] # # COUNT # if (is_method("BYE")) { $var(n) = $shtdec(callcount=>active); sht_rm_name_re("callcount=>$ft"); } -------------------------------------------- # Manage incoming replies onreply_route[MANAGE_REPLY] { # # COUNT # if (status=="200") { if (is_method("INVITE")) { # On start if($sht(callcount=>active) == $null) { $sht(callcount=>active) = 0; } # New call if ($sht(callcount=>$ft::from_tag)== $null) { $sht(callcount=>$ft::from_tag) = $ft; $var(n) = $shtinc(callcount=>active); } } xlog("L_INFO", "This is a 200 OK reply\n"); } xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } ------------------------------------------------------- However I wanted to distribute calls to several servers I further added the despatcher code below # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "-----RELAYING going to <$ru> via <$du> time [$Tf] -----\n"); This makes my counting code above give incorrect call count. How can I make call count work with dispatcher ? Do advise how I can do this Regards

2 1

Create "Contac Groups" on the server Kamailio
by ALFONSO MARTINEZ GOMEZ 24 May '16

24 May '16

Hi, I'm working on a VoIP communication application with Kamailio as SIP server and using the SIREMIS interface. As an administrator, I can add "Subscribers" which later will be registered on the server so that they can make the necessary negotiations when they want to make a call. My question is, how you can make groups of "contacts" so that such calls can be made only with each other "contacts" in the same group? that is, we can have many "subscribers" on the server and belong to two different groups and that they can only talk to each other contact each group. Thank you so much and greetings!

1 0

doubt using rate limit
by david escartin 24 May '16

24 May '16

Hello all i have a quick question about using module ratelimit on kamailio 4.2 or 4.4. on tests made before running it on production, i set the RED algorithm with a limit of 10 calls per second, but i saw the call limitation was not done until we got more than 20 calls per second or so on the queue. we didnt get any INVITE discarded. after surpassing around 20 INV/sec, it's true the algorithm starts to work fine and only 10INV/sec are passed more or less on production i have no easy way to check this behaviour, besides it's a feature barely used. so i only want to know if this behaviour is known, or if i'm missing soemthing my config is like this modparam("ratelimit", "timer_interval", 5) modparam("ratelimit", "queue", "4:INVITE") modparam("ratelimit", "pipe", "4:RED:10") route[RATE_LIMIT] { $var(limitation)="4"; if (!rl_check("$var(limitation)")) { send_reply("505","Limiting"); xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, rejecting with 505\n"); route(CLEAR); exit; } } thanks a lot and regards david escartin

2 1

Please help me setting up Kamailio for android application
by Kunal Dhanda 24 May '16

24 May '16

I wanted to create my own voip application using kamailio . I am not able to find any rest integration with kamailio for my server to use for user registration and other task.Please help in developing my appplication. -- Thanks & Regards, Kunal Dhanda, Software Developer E-mail : < "kunaliitkgp09(a)gmail.com" >, Ph. No. (M) : +91 - 8884818805

2 1

Found bug at ACC
by Pavel Eremin 24 May '16

24 May '16

Hi all, when we set syslog facility for ACC to LOCAL3, log_write_cdr() will write at LOG_DAEMON anyway. You have to recompile module acc to user another default syslog facility to save cdr with another syslog facility.

2 1

kamctl stats shmem error changing shared memory
by Mr.Gab 23 May '16

23 May '16

Hi, i have a problem with my kamailio installation. Actually i set the shared memory to 128MB and it's all fine. When i increase the shared memory to 256Mb and more i get the following error while checking kamctl commands (like kamctl stats shmem) ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. If i revert to 128Mb kamailio is able to create the kamailio_fifo file and all works fine. Can somebody help me to solve? Thanks

3 3

Re: [SR-Users] Change CANCEL reason when fr_timer expires
by Bruno Emer 22 May '16

22 May '16

Hello, Alex! Fisrt of all, thanks for you help with this problem! I don't think that this will help me. I can see that this parameter is enabled by default and in fact it is working in some cases. One case where this works is when the INVITE is sended for more than one endpoint (when the same user is registered in many devices). When someone answers the INVITE, Kamailio sends a CANCEL message to the other endpoints and it has the Reason header showing that the call was answered elsewhere. I want to add the CANCEL Reason header when the timer expires, so kamilio will not receive a final replay to that transaction. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-20 15:42 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > I have a situation here where when I send a INVITE to some specific > user-agents in my environment I must change the fr_timer to a value lower > than the default. > > I can acomplish this using the "t_set_fr" commands in my cfg file, and it > works like a charm. > > The point is that when the timer reaches its limit, the kamailio server > sends a CANCEL message to the user-agent. I know that this is the normal > behavior, but I want to add the Reason field on it, so the user-agent > application can know that the call wasn't answered on time. > > I was looking the TM module documentation to understand how to do this, > but I couldn't figure out a way to get this working. > > Is there a way to add the Reason in CANCEL messages generated by the > fr_timer? > > > Thanks. > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >

2 1

Change CANCEL reason when fr_timer expires
by Bruno Emer 21 May '16

21 May '16

Hello all! I have a situation here where when I send a INVITE to some specific user-agents in my environment I must change the fr_timer to a value lower than the default. I can acomplish this using the "t_set_fr" commands in my cfg file, and it works like a charm. The point is that when the timer reaches its limit, the kamailio server sends a CANCEL message to the user-agent. I know that this is the normal behavior, but I want to add the Reason field on it, so the user-agent application can know that the call wasn't answered on time. I was looking the TM module documentation to understand how to do this, but I couldn't figure out a way to get this working. Is there a way to add the Reason in CANCEL messages generated by the fr_timer? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

2 1

extra characters in re-invite "TO" field
by Al S 20 May '16

20 May '16

Hi, Kamailio adds extra characters in the re-invite "TO" field during session timer. There is no header manipulation in our route[WITHINDLG] block. Re-INVITE received by KamailioTo: <sip:8131111111@?.?.?.?;otg=1>;tag=1c1048632956 Re-INVITE forwarded by KamailioTo: <sip:8132222222@?.?.?.?;T\033\023Z\f1>;tag=1c1048632956 Your help would be greatly appreciated. Thanks,AS

2 2

Question about testing RTPENGINE
by Dmitry 20 May '16

20 May '16

Hello I am trying to make a test setup to test the RTPENGINE. Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.132)-----via NAT-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) First WITHOUT RTPENGINE : When A calls to B - the sound both ways, so it is ok. Second WITH RTPENGINE: When A calls to B - the sound both ways, so it is ok. Actually what I want is to make such setup that calls without the RTPENGINE are impossible.

1 0

WebRTC
by Moacir Ferreira 20 May '16

20 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

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Re: [SR-Users] Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 19 May '16

19 May '16

Is there is reason why this post does not get approved? From: Pejman Rajabian [mailto:pejman@italkip.com] Sent: Wednesday, May 18, 2016 12:59 PM To: 'sr-users(a)lists.sip-router.org' Subject: Kamailio notify conflict when a second call lands on a phone I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 pejman(a)italkip.com

2 1

Sipcapture on SPARC
by Spencer Thomason 18 May '16

18 May '16

Hello, I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing some strange date related errors: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '2052-06-12 16:21:06' for column 'date' at row 1 (1292) Is there some sore of endianness issue? Thanks for any assistance! Regards, Spencer

2 3

[kamailio 5.0] upgrade VS rollback/backward compatibility
by Alex Lutay 18 May '16

18 May '16

Hi, We are heavily using Kamailio in our product Sipwise NGCP and currently testing possibility to provide rollback for the customers with the huge systems as the last resort functionality in case if upgrade went wrong. We hope never use it on practice but prefer to have such possibility. The main logic here is to change DB in 'backward compatible' way. So OLD code must work with NEW DB schema (but NOT a vice verse). We have some rules which helps us care about it, like "Do not drop columns, just discontinue them" and "All new fields should have defaults", etc. Also we have separated 'code', 'configs' and 'data' parts of our product and rollback 'code'/'configs' only, while 'data' (e.g. MySQL DB) stays in new state. Otherwise rollback will require a lot if time to revert all DB statements and sometimes it is just not possible. While every time we have to upgrade Kamailio we have a problems with rollback possibility. The point here that Kamailio checks DB table_version and doesn't start if the version mismatch found. Q: What about to add Kamailio possibility to work in backward compatible mode? Sample for upgrade Kamailio 4.3.x -> 4.4.x (the versions here as an example only!) https://www.kamailio.org/wiki/install/upgrade/4.3.x-to-4.4.0 We have 3 DB statements there which are adding new columns to 3 tables, like > -- table: active_watchers > ALTER TABLE active_watchers > ADD COLUMN flags INT(11) NOT NULL DEFAULT '0', > ADD COLUMN user_agent VARCHAR(255) NOT NULL; > DELETE FROM version WHERE TABLE_NAME='active_watchers'; > INSERT INTO version (`table_name`, `table_version`) VALUES ('active_watchers','12'); Problem here is missed default for column 'user_agent' which can be empty value as for me. So, lets assume we have it there. Then we can add new column to table 'version', like that: > mysql> select * from version where table_name='active_watchers'; > +-----------------+---------------+---------------------+ > | table_name | table_version | table_compatibility | > +-----------------+---------------+---------------------+ > | active_watchers | 12 | 11 | > +-----------------+---------------+---------------------+ > 1 row in set (0.00 sec) Currently: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 Proposed behaviour: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 and compatibility option is not set - if compatibility option is set, check value of table_compatibility and refuse to load if current version less then table_compatibility value >From the sample above I would say module presence in Kamailio 4.4.x is compatible with Kamailio 4.3.x if we define default value for the field 'user_agent'. Having that will allow us to rollback on previous NGCP release and _temporary_ work on Kamailio 4.3.x having new Kamailio DB 4.4.x P.S. This is just a question to check Kamailio community vision here and reality is more complex as described, so some additional checks/limitations will be in place. P.P.S Compatibility Kamailio option can/should be disabled by default. Field 'table_compatibility' can be outside table 'version', e.g. in separate table. Complete compatibility logic can be some separate Kamailio 'compatibility' module. Thank you! -- Alexander Lutay Head of Quality Assurance Sipwise GmbH, Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Office: +43(0)13012036 Email: alutay(a)sipwise.com Website: https://www.sipwise.com Meet us @ ANGACOM: Hall 10.1 / booth N10 Exhibition and Congress for Broadband, Cable & Satellite: 07 – 09 June, 2016 in Cologne

4 3

Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 18 May '16

18 May '16

I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 <mailto:pejman@italkip.com> pejman(a)italkip.com

1 0

invalid route error when turning on Roles in kamailio kazoo
by Dmitry 18 May '16

18 May '16

hello When I uncomment in /etc/kazoo/kamailio/local.cfg#!trydef TRAFFIC-FILTER-ROLE (it was commented previously) And the kamailio does not start and the log gives this error:-Invalid route satement The structure of files is simple:the trafic-filter-role is in the separate file. and in the main file (default.cfg #!ifdef TRAFFIC_FILTER-ROLEinclude_file "traffic-filter-role.cfg" and that's it. But it gives INVALID ROUTE ERROR

2 2

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 17 May '16

17 May '16

Hello, David. Sorry for the time to reply you. I was a little busy here in the last few days. I am not sure about the "==". We had tried it here, but it only worked when we used with =~. We are currently running kamailio 4.2, so it may be a problem with our version. About the $hdr variable, it is used to get information from the SIP Headers. I am not sure, but I dont think that there is an header with Instance-ID, so you must try to find how can you get this ID in kamailio, as I don't know where is it saved. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-13 12:24 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > After some time I was able to get this working here, and now I am coming > back to share my configuration with you guys (in case someone needs to get > this working). > > Here is what we have done: > > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > > xlog("L_INFO", "checking $tu registered with > User-Agent:$hdr(User-Agent)\n"); > if(reg_fetch_contacts("location", "$tu", "caller")) > { > xlog("L_INFO", "$(ulc(caller=>count)) records > found for $tu\n"); > $var(i) = 0; > while($var(i) < $(ulc(caller=>count))) > { > > xlog("L_INFO", "record $var(i) with > ruid:$(ulc(caller=>ruid)[$var(i)]) - > User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); > if > ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) > { > $var(ruid) = > $(ulc(caller=>ruid)[$var(i)]); > xlog("L_INFO", "unregistering $tu > ruid:$var(ruid)"); > unregister("location", "$tu", > "$var(ruid)"); > } > $var(i) = $var(i) + 1; > } > } > if (!save("location", "0x00")) > sl_reply_error(); > > exit; > } > } > > > There are some logs over there because we used it to see what was going on > while testing the configurations. > > > With this we can get just one register per User-Agent. > > > And again, thanks Olle for your help in this problem > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Sure it is! >> >> This is exactly what I was looking for! Actually I wasn't able to see >> that there was a way to perform this action! >> >> Thanks for your help, Olle! >> >> I will try to create this, and when I have it done, I will post my script >> here, so if someone else need the same problem I can help. >> >> Now, I have just one more question about my scenario: when I register a >> user, is there a way to create something like a "custom field" on the >> location? To explain better: If possible, I can add something like a custom >> header that says if the user is registered from a phone device or the web >> interface, then I can perform the loop and replace only the oldest contact >> using that device... >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> This is my problem... I want to have 2 contacts for AOR, and not only >>> one. >>> >>> When the 3rd arrives, it must remove the oldest and continue with 2... >>> >>> *If I am using the 0x04 parameter, it will save just one contact, and >>> the user will not be able to get the Invite both, in the mobile device and >>> web interface.* >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >>> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello, *Olle!* >>>> >>>> *Thanks for your help in this case, and don't worry about the time >>>> (actually I wasn't even expecting to get an answer today).* >>>> >>>> >>>> *I know the max_contacts parameter. But the problem is that if I set >>>> the max_contacts to 2, when the user tries to register again, he gets an >>>> error message 503. The point is that I want to get the user registered, and >>>> replace the oldest location entry for the newest one.* >>>> >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>>> >>>>> Hello all. >>>>> >>>>> I have problem here and I tried to find a solution and search over >>>>> internet, but without success. >>>>> >>>>> My scenario is the following: I have an application that must be >>>>> registered in Kamailio when a user logs in the web interface, so he can get >>>>> calls (something like a web softphone using webrtc). At this point, we are >>>>> OK, and everything is working fine. >>>>> >>>>> To get these register functions I am using the parameter >>>>> "save("location", "0x04")" as described in the REGISTRAR module >>>>> documentation, so if a user logs in another web browser or computer, only >>>>> the last one will continue registered and all calls will be forwarded to >>>>> him. >>>>> >>>>> The point is that now we are creating a phone app that will do almost >>>>> the same thing as the web interface, allowing users to receive calls using >>>>> the mobile device, and here is my problem: I want to allow my users to be >>>>> registered on two devices at the same time, but if a user logs into another >>>>> device, I don't want to reply with a 503. I want to allow the user to >>>>> register again, deregistering the oldest contact. >>>>> >>>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>>> description it says "implemented overwrite oldest contact behaviour", >>>>> but I couldn't find any documentation about it. >>>>> >>>>> So, is there a way to get this working today? >>>>> >>>>> >>>>> *Bruno Emer* >>>>> >>>>> Mobile: +55 11 96540-0044 >>>>> email: brunoemer(a)gmail.com >>>>> >>>> >>>> >>> >> >

1 0

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Enum module and parallel branching
by Loic Chabert 17 May '16

17 May '16

Hello, I'm trying to use enum module at core level to route my numbers to end-point asterisk servers. As i read (and validated by tests), when a phone number has two DNS NAPTER entries, enum module create two branch, one for each entries, and send request to asterisk servers (in the same time). Do you know any tricks or functions to disable parallel branching, and make serial branching instead (following responses returned by asterisk) ? I have also change preference and order in dns config file, same results. Thanks in advance. Regards.

2 3

MySQL database for kamailio as S-CSCF
by Christoph Valentin 17 May '16

17 May '16

Hi Community, I'm doing very first steps to use kamailio as an S-CSCF. Now I'm confused about some defines in the examples/scscf/scscf.cfg file. #!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" Tutorials/ims/Installation-howto tell me only I-CSCF needs a database, I do not find anything about database for S-CSCF. However, when I comment out the defines ##!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" then the kamctl start results in a segmentation fault. I have installed kamailio-4.4.0 from RPMs. Is there any hint about database for S-CSCF? Thanks in advance Christoph

1 1

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by M R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Debian, dnssec.so, and missing dnsval.conf
by David Holl 17 May '16

17 May '16

In the interest of sharing and promoting DNSSEC, I was recently stumped on a problem where I noticed I could not call sip:test.echo@sip5060.net without disabling the dnssec.so module in my kamailio.cfg. The last time I had tried the sip5060 echo test was several months ago, and in the intervening time, I had applied the numerous debian package updates. Thus, I could not isolate the cause of the problem to one of the many package updates such as kamailio 4.4.x, libval, my own DNSSEC dns server, etc... My initial only clue was kamailio's syslog messages: May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve "sip5060.net" :unresolvable A or AAAA request (-7) May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [t_fwd.c:1712]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) To summarize the problem, I found that kamailio's dnssec.so was failing queries to domains that were not DNSSEC protected such as sip5060.net. This behavior is a bug, since sip5060.net is in this case, provably not participating in DNSSEC from the toplevel "net." zone [1]. The fix was to provide a default configuration file to libval, which was somehow not included in Debian's libval15 package. I created the following configuration from the default dnsval.conf in the source distribution: https://dnssec-tools.org/download/dnsval-2.2.tar.gz https://dnssec-tools.org/download/dnsval-2.2.tar.gz.asc as well as their wiki: https://www.dnssec-tools.org/wiki/index.php/Dnsval.conf cat << 'EOF' > /etc/dnssec-tools/dnsval.conf ################################## # Global Options ################################## global-options trust-oob-answers yes # I'm running a DNSSEC nameserver locally, # so 4 KiB queries to localhost should be OK: edns0-size 4096 env-policy disable app-policy disable rec-fallback no ; : trust-anchor . DS 19036 8 2 49AAC11D7B6F6446702E54A1607371607A1A41855200FD2CE1CDDE32F24E8FB5 ; : zone-security-expectation . validate ; : provably-insecure-status . trusted ; : clock-skew . 0 ; EOF In summary: I find it odd that a default dnsval.conf was not included with my system's normal installation of the "libval15:amd64" package, version 2.2-4. After providing dnsval.conf, kamailio's dnssec.so was happy again. Side notes: * To identify the directory for dnsval.conf, run libval-config --dnsval-config On Debian, libval-config is in the libval-dev pacakge, and this package includes a man page for dnsval.conf as well. * To test your libval and dnsval.conf, try dt-validate -s On Debian, dt-validate is in the libval-bin package. - David [1] -- Authenticated denial of existence: https://tools.ietf.org/html/rfc5155 https://tools.ietf.org/html/rfc7129 And for sip5060.net, there's an NSEC3 record in the top level net. zone proving non-existance of sip5060.net DS records: http://dnsviz.net/d/sip5060.net/dnssec/ Translation: sip5060.net provably not enabled DNSEC.

2 1

kamailio not authenticate a subscribed username
by Al S 16 May '16

16 May '16

Hi,1) A generic subscriber is added to kamailio database:kamctl add pbx password2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call:UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it)The correct password is being sent to kamailio.Your assistance is greatly appreciatedThanks,AR

1 0

kamailio not authenticate a call with subscribed username
by Al S 16 May '16

16 May '16

Hi, 1) A generic subscriber is added to kamailio database:kamctl add pbx password 2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call: UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it) The correct password is being sent to kamailio. Your assistance is greatly appreciated Thanks,AR

1 0

Re: [SR-Users] Another question about RTPengine module
by Dmitry 16 May '16

16 May '16

the problem is: rtpengine was compiled with USE_SCTP,but kamailio that was installed via RPM is without this flag. How to tell compiler NOT TO " USE USE_SCTP" flag? +++++++++++++++++++++++ Makefile.defs:SCTP ?= 0 - and it compiles without sctp. Now kamailio.log:ERROR: rtpengine [rtpengine.c:1113]: child_init(): Name or service not knownERROR: <core> [sr_module.c:929]: init_mod_child(): Error while initializing module rtpengine (/usr/lib64/kamailio/modules/rtpengine.so) View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ 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 On Sunday, May 15, 2016 6:21 PM, Dmitry <mbike2000ru(a)yahoo.com> wrote: yes, I wrote that modparam are recognized.MODPARAM must be placed i the sae place where the RTPPROXY resides in the config.+++++++++++++++++++++++++++++++ Another question:kazoo kamailio KAMAILIO-V :4.3.5:kamailio -vversion: kamailio 4.3.5 (x86_64/linux)flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknowncompiled on 17:51:55 Mar 8 2016 with gcc 4.4.7 To add the RTPENGINE - I found Kamailio 4.3.5 sources https://www.kamailio.org/pub/kamailio/4.3.5/src/ When I start kamailio I get the following error:ERROR: <core> [sr_module.c:431]: version_control(): module compile flags mismatch for /usr/lib64/kamailio/modules/rtpengine.so I did the following to add missing modules: #make include_modules="rtpengine topoh" cfg #make all cp /usr/src/kamailio-4.3.5/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so How to fix it? The sources version seems to be the same On Saturday, May 14, 2016 1:21 AM, ycaner <yasin.caner(a)netgsm.com.tr> wrote: Hello; i am not sure. i downloaded from git kamailio. add rtpengine instead of rtpproxy . it works fine. it is hard to solve problem without log. Cheers. try to run with kamailio -E -dd , maybe it gives information. check kamailio.cfg is that rtpengine is loaded like loadmodule "rtpengine.so" 2016-05-13 11:35 tarihinde Dmitry [via SIP Router] yazdı: one more question about rtpengine. I installed Kamailio (centos 6.7) from source as You recommended make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg no rtpengine daemon (it is not installed) but the picture is the same: modparam ("rtpengine,....) kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(19523) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found it means that it does not depend upon rtpengine daemon. # single rtproxy modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") I copied it from http://www.kamailio.net/docs/modules/4.2.x/modules/rtpengine.html STRACE kamctl start -says nothing anout rtpengine What else is it possible to do to find the cause of the problem? On Friday, May 13, 2016 11:45 AM, ycaner <[hidden email]> wrote: Hello; rtpengine module isnt default. you need to add with "make include_modules". after make && make install , automatically it will be copied to lib/modules. Cheers. 2016-05-13 09:42 tarihinde Dmitry [via SIP Router] yazdı: And one more question on it: If the Kamailio was installed from RPM and rtpengine.so is missing - May I add rtpengine.so the following way: make include_modules="rtpengine" cfg make all and then copy rtpengine.so to /modules/ directory Is this way (adding missing modules) is correct? On Friday, May 13, 2016 11:25 AM, Dmitry <[hidden email]> wrote: hello I tested on Centos 6.7 as well as Ubuntu server - the problem is the same. RTPENGINE was installed both from dpkg and from Source (manual compilation) So I need a clarification on it. RTPENGINE daemon is started. The Kamailio module itself is taken from /usr/src/kamailio, where the source was compiled. mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so sudo /etc/init.d/kamailio restart | grep rtp ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <rtpengine> found version: kamailio 5.0.0-dev4 (x86_64/linux) Any help is appreciated. On Thursday, May 12, 2016 6:04 PM, Dmitry <[hidden email]> wrote: Hi, I installed Kamailio from GIT. I did make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg Actually loadmodule "rtpengine.so" does not affect kamailio start/restart. modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") causes this failure: [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found On Thursday, May 12, 2016 12:26 PM, ycaner <[hidden email]> wrote: Hello; on bash , kamailio -V gives Version , kamailio -c gives debug result. you can ask from here https://groups.google.com/forum/#!forum/2600hz-dev for kazoo rpm. if there is something wrong , they will fix it. you can install from source code from here => https://www.kamailio.org/w/ and "make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg " in kamailio folder before 'make all' , helps you add modules. if there is a dependecies is missing , gives error. Cheers. 2016-05-11 19:51 tarihinde Dmitry [via SIP Router] yazdı: I installed Kamailio : yum install -y kazoo-kamailio https://2600hz.atlassian.net/wiki/display/Dedicated/via+RPM git clone https://github.com/sipwise/rtpengine.git the latest version of rtpengine and I do not know how to view the version of Kazoo-Kamailio On Wednesday, May 11, 2016 6:31 PM, ycaner <[hidden email]> wrote: hello ; maybe some devel are missing. which version using?i will try. cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML --

1 4

Another question about RTPengine module
by Dmitry 15 May '16

15 May '16

I will clarify the question in my previous post.I installed Kamailio from 2600hz repo via yum (it is for kazoo). But when cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so the kamailio does not see this moduleNo module matching <rtpengine> found Is there something like modules.list (it is used when I compile Kamailio manually) when the kamailio was installed via RPM?

2 12

WebRTC integration
by Moacir Ferreira 15 May '16

15 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

1 0

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 13 May '16

13 May '16

Hello all! After some time I was able to get this working here, and now I am coming back to share my configuration with you guys (in case someone needs to get this working). Here is what we have done: route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } xlog("L_INFO", "checking $tu registered with User-Agent:$hdr(User-Agent)\n"); if(reg_fetch_contacts("location", "$tu", "caller")) { xlog("L_INFO", "$(ulc(caller=>count)) records found for $tu\n"); $var(i) = 0; while($var(i) < $(ulc(caller=>count))) { xlog("L_INFO", "record $var(i) with ruid:$(ulc(caller=>ruid)[$var(i)]) - User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); if ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) { $var(ruid) = $(ulc(caller=>ruid)[$var(i)]); xlog("L_INFO", "unregistering $tu ruid:$var(ruid)"); unregister("location", "$tu", "$var(ruid)"); } $var(i) = $var(i) + 1; } } if (!save("location", "0x00")) sl_reply_error(); exit; } } There are some logs over there because we used it to see what was going on while testing the configurations. With this we can get just one register per User-Agent. And again, thanks Olle for your help in this problem *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Sure it is! > > This is exactly what I was looking for! Actually I wasn't able to see that > there was a way to perform this action! > > Thanks for your help, Olle! > > I will try to create this, and when I have it done, I will post my script > here, so if someone else need the same problem I can help. > > Now, I have just one more question about my scenario: when I register a > user, is there a way to create something like a "custom field" on the > location? To explain better: If possible, I can add something like a custom > header that says if the user is registered from a phone device or the web > interface, then I can perform the loop and replace only the oldest contact > using that device... > > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > > 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> This is my problem... I want to have 2 contacts for AOR, and not only >> one. >> >> When the 3rd arrives, it must remove the oldest and continue with 2... >> >> *If I am using the 0x04 parameter, it will save just one contact, and the >> user will not be able to get the Invite both, in the mobile device and web >> interface.* >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> <https://www.facebook.com/bruno.emer.5> >> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >> <https://twitter.com/brunoemer_> >> <https://www.flickr.com/photos/122070309@N03/> >> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >> >> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello, *Olle!* >>> >>> *Thanks for your help in this case, and don't worry about the time >>> (actually I wasn't even expecting to get an answer today).* >>> >>> >>> *I know the max_contacts parameter. But the problem is that if I set the >>> max_contacts to 2, when the user tries to register again, he gets an error >>> message 503. The point is that I want to get the user registered, and >>> replace the oldest location entry for the newest one.* >>> >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello all. >>>> >>>> I have problem here and I tried to find a solution and search over >>>> internet, but without success. >>>> >>>> My scenario is the following: I have an application that must be >>>> registered in Kamailio when a user logs in the web interface, so he can get >>>> calls (something like a web softphone using webrtc). At this point, we are >>>> OK, and everything is working fine. >>>> >>>> To get these register functions I am using the parameter >>>> "save("location", "0x04")" as described in the REGISTRAR module >>>> documentation, so if a user logs in another web browser or computer, only >>>> the last one will continue registered and all calls will be forwarded to >>>> him. >>>> >>>> The point is that now we are creating a phone app that will do almost >>>> the same thing as the web interface, allowing users to receive calls using >>>> the mobile device, and here is my problem: I want to allow my users to be >>>> registered on two devices at the same time, but if a user logs into another >>>> device, I don't want to reply with a 503. I want to allow the user to >>>> register again, deregistering the oldest contact. >>>> >>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>> description it says "implemented overwrite oldest contact behaviour", >>>> but I couldn't find any documentation about it. >>>> >>>> So, is there a way to get this working today? >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>> >>> >> >

2 1

Changing the SDP connection-address to a value in a pseudovariable
by Kevin McAllister 13 May '16

13 May '16

counter to the docs for nathelper fix_nated_sdp the config parser rejects any pseudovariables in the second parameter claiming they need to be constant. Does anyone know another way to change the "c=" line in an SDP? Or do I have to patch the nathelper module to include a free_fixup_function for the fix_nated_sdp stuff? This is on kamailio 4.2 Thanks, Kevin

1 1

Kamailio Crash
by Sunil More 13 May '16

13 May '16

Hello All, Kamailio instance just crashed with the following logs May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12675]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12674]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12682]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12684]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12667]: INFO: <core> [mem/f_malloc.c:598]: fm_free(): freeing a free fragment (0x7f23ff39cb88/0x7f23ff39cbd0) - ignore to add i have collected a gdb backtrace as below, root@IP:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/c/core.kamailio.sig11.12671 chetan/ corefiles/ root@P172:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/c/core.kamailio.sig11.12671 chetan/ corefiles/ root@P172:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/corefiles/core.kamailio.sig11.12671 The version of Kamilio used by me is as below;. version: *kamailio 4.4.1 (x86_64/linux) 90be8b* flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 90be8b Thanking You, Sunil More Ph : 9503338275

2 2

Blacklists and DB_Cassandra
by SamyGo 13 May '16

13 May '16

Hi, I am tasked to make use of blacklist module for about 4 to 6 million numbers. I am thinking of using Cassandra for the purpose but reading through the documentatiom of module and recent mailing list discussion made me a bit hesitant. I am looking for advise on this whether this is going to perform as expected or is not even going to work with the BlackListing module ! I am using Cassandra 33x and Kamailio 4.4 over Ubuntu 14.04. Looking for suggestions here. Regards, Sammy

2 4

DB_Cassandra + UserBlacklist = ERROR
by SamyGo 12 May '16

12 May '16

Hi List, I'm trying to hook up userblacklist module with db_cassandra. I've kamailio keyspace configured with tables for userblacklist , globalblacklist, and version are created with some data in there. Here is setup info: version: kamailio 4.4.1 *kamailio.cfg* ... loadmodule "db_cassandra.so" modparam("db_cassandra", "schema_path","/etc/kamailio/kamailio") loadmodule "userblacklist.so" modparam("userblacklist", "db_url", "cassandra://:@127.0.0.1:9160/kamailio") modparam("userblacklist", "userblacklist_table", "userblacklist") modparam("userblacklist", "globalblacklist_table", "globalblacklist") *SCHEMA PATH:* root@whit-list:/etc/kamailio/kamailio# ls userblacklist version root@whit-list:/etc/kamailio/kamailio# cat version/version table_name(string) table_version(int) table_name root@whit-list:/etc/kamailio/kamailio# cat userblacklist/userblacklist id(int) username(string) domain(string) prefix(string) whitelist(int) id username *CASSANDRA DB* root@whit-list:/etc/kamailio/kamailio# cqlsh Connected to Test Cluster at 127.0.0.1:9042. [cqlsh 5.0.1 | Cassandra 3.5 | CQL spec 3.4.0 | Native protocol v4] Use HELP for help. cqlsh> use kamailio ... ; cqlsh:kamailio> describe tables; globalblacklist version userblacklist cqlsh:kamailio> select * from version; table_name | table_version -----------------+--------------- uacreg | 2 version | 1 globalblacklist | 1 userblacklist | 1 (4 rows) Every time I start up Kamailio I see the following debug logs: 0(13877) DEBUG: db_cassandra [dbcassa_table.c:572]: dbcassa_read_table_schemas(): Full name= /etc/kamailio/kamailio/ 0(13877) DEBUG: db_cassandra [dbcassa_table.c:589]: dbcassa_read_table_schemas(): Full dir name= /etc/kamailio/kamailio/version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:609]: dbcassa_read_table_schemas(): Found database version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:620]: dbcassa_read_table_schemas(): database table version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:630]: dbcassa_read_table_schemas(): File path= /etc/kamailio/kamailio/version/version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:200]: dbcassa_load_file(): loading file [/etc/kamailio/kamailio/version/version] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:122]: dbcassa_table_new(): mtime is 1463068672 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [table_name] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[0] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[0] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [table_version] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[1] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[1] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [table_name] in primary key 0(13877) DEBUG: db_cassandra [dbcassa_table.c:589]: dbcassa_read_table_schemas(): Full dir name= /etc/kamailio/kamailio/userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:609]: dbcassa_read_table_schemas(): Found database userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:620]: dbcassa_read_table_schemas(): database table userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:630]: dbcassa_read_table_schemas(): File path= /etc/kamailio/kamailio/userblacklist/userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:200]: dbcassa_load_file(): loading file [/etc/kamailio/kamailio/userblacklist/userblacklist] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:122]: dbcassa_table_new(): mtime is 1462985556 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [id] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:258]: dbcassa_load_file(): column[0] is INT! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [username] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[1] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[1] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [domain] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[2] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[2] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [prefix] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[3] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[3] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [whitelist] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:258]: dbcassa_load_file(): column[4] is INT! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [id] in primary key 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [username] in primary key ..... 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:149]: db_cassa_new_connection(): opening connection: cassa:// xxxx:xxxx@127.0.0.1:9160/kamailio 0(12780) DEBUG: <core> [mem/f_malloc.c:439]: fm_malloc(): fm_malloc(0x7f4e36140010, 48) called from db_cassandra: dbcassa_base.cpp: db_cassa_new_connection(155) 0(12780) DEBUG: <core> [mem/f_malloc.c:514]: fm_malloc(): fm_malloc(0x7f4e36140010, 48) returns address 0x7f4e361bb5f0 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:161]: db_cassa_new_connection(): 0x7f4e361bb5f0=pkg_malloc(48) 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:117]: dbcassa_open(): Opened connection to Cassandra cluster 127.0.0.1:9160 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:955]: db_cassa_query(): query table=version 0(12780) ERROR: db_cassandra [dbcassa_base.cpp:449]: *cassa_translate_query(): table version does not exist!* 0(12780) ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster 0(12780) ERROR: <core> [db.c:397]: db_table_version(): error in db_query 0(12780) ERROR: <core> [db.c:436]: db_check_table_version(): querying version for table userblacklist 0(12780) ERROR: userblacklist [db_userblacklist.c:86]: userblacklist_db_init(): during table version check. 0(12780) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(12780) DEBUG: <core> [mem/f_malloc.c:575]: fm_free(): fm_free(0x7f4e36140010, 0x7f4e361bb5f0), called from db_cassandra: dbcassa_base.cpp: db_cassa_free_connection(203) 0(12780) DEBUG: <core> [mem/f_malloc.c:594]: fm_free(): fm_free: freeing block alloc'ed from db_cassandra: dbcassa_base.cpp: db_cassa_new_connection(155) 0(12780) DEBUG: <core> [mem/f_malloc.c:575]: fm_free(): fm_free(0x7f4e36140010, 0x7f4e361bb318), called from core: db.c: db_do_close(356) 0(12780) DEBUG: <core> [mem/f_malloc.c:594]: fm_free(): fm_free: freeing block alloc'ed from core: db.c: db_do_init2(298) 0(12780) ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module userblacklist (/usr/local/lib64/kamailio/modules/userblacklist.so) ERROR: error while initializing modules Any help would be highly appreciated. Regards, Sammy

2 2

route_if_exists crash / breakage
by Alex Balashov 12 May '16

12 May '16

Hi, I'm running 4.4.0:ea2033 and am seeing a crash when using route_if_exists() on a nonexistent route: Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m 512 -M 8 -u evaristesys -g'. Program terminated with signal 11, Segmentation fault. #0 0x0000000000456751 in do_action (h=0x7ffcce5bc4c0, a=0xf0f0f0f0, msg=0x7fba440abc80) at action.c:337 337 switch ((unsigned char)a->type){ gdb) print a $1 = (struct action *) 0xf0f0f0f0 (gdb) print a->type Cannot access memory at address 0xf0f0f108 The invocation looks like this: if(check_route_exists("AUX_RTPENGINE_OFFER_INBOUND")) route_if_exists("AUX_RTPENGINE_OFFER_INBOUND"); else { set_rtpengine_set("1"); rtpengine_offer("replace-origin replace-session-connection ICE=remove"); } #route[AUX_RTPENGINE_OFFER_INBOUND] { # xlog("L_INFO", 'action=R-AUX-RTPENGINE-OFFER-INBOUND'); #} Uncommenting the nonexistent route has another curious effect: script execution appears to break after route[AUX_RTPENGINE_OFFER_INBOUND] executes, and nothing further happens. Changing route_if_exists("AUX_RTPENGINE_OFFER_INBOUND"); to route(AUX_RTPENGINE_OFFER_INBOUND); resolves both issues (the crash and the script execution halting). But, of course, if the route is commented out - its normal state - then Kamailio won't start since it won't allow a nonexistent identifier to be passed to route(). -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 6

Kamailio - Manage several kamailio instances with Siremis
by José Seabra 12 May '16

12 May '16

Hi list, Anyone here knows if i can use one siremis instance to manage more than one Kamailio Instance? BR -- Cumprimentos José Seabra

1 0

Kamailio / issue with registration
by Ali Taher 12 May '16

12 May '16

Hello, I installed Homer with kamailio as capture agent in order to capture sip packets. The first test I did was using the default kamailio.cfg config located in homer-docker directory , where all Register and Invite packets were saved in the database and I was able to monitor them from web homer. After that , I tried to enable registration using the save(location) function in registrar module , yet I'm getting the below error in the log : 2016-05-11T18:48:36.443231+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f64702b1d98,767,0,192.168.26.3:3806,16): Bad file descriptor(9) 2016-05-11T18:48:36.443239+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: sl [../../forward.h:201]: msg_send_buffer(): udp_send failed 2016-05-11T18:48:36.443247+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: registrar [reply.c:734]: reg_send_reply(): failed to send 200 OK Can you please help with resolving this issue , noting that the registration works good if I remove the routing logic set for sipcapture. Any hint would be appreciated. Cheers, Ali Taher

1 1

Wiki update notice
by Mikko Lehto 12 May '16

12 May '16

Hello Some wiki work: 1. "What's new in 4.4" -content is no more present in devel version wiki page 2. based on recent mailing list discussion, I added note about github pull requests I think these make sense, please yell if not :) [1] http://kamailio.org/wiki/features/new-in-devel [2] http://kamailio.org/wiki/devel/github-contributions?rev=1456916916&do=diff -- Mikko

2 2

T_RELAY_TO distribute local proxy to a central one
by Sébastien Brice 12 May '16

12 May '16

Hi guys, i have a kamailio running on the wan (my central kamailio) and i have phones on a private network range 172.20.0.0/16 behind a NAT. i also run another kamailio locally to collect all sip phones on my network and this one use the DB of the remote central kamailio. the local kamailio does t_relay_to() the ip of the central kamailio route { if (uri==myself) { if (method=="REGISTER") { route(2); exit; }; route[2] { rewritehost("public_remote_kamailio"); if(!t_relay_to("udp:public_remote_kamailio:5060")) { sl_reply_error(); }; exit; } the bad thing is once the phones get registered with non-routable ip (like in the attached picture) trying to INVITE them directly by requesting public_kamailio_ip result in non-working situation. how should I rewrite my dst_domain when registring in a way INVITE triggers to the local kamailio. My local kamailio runs with the listen udp:private_local_kamailio advertise udp:public_local_kamailio I try to be explicit but my understanding is not perfect, it would be nice you could give me a hand. I put my local kamailio.cfg attached. thx

2 1

4.4 compiler warnings on debian jessie
by Juha Heinanen 12 May '16

12 May '16

debian jessie c compiler complains about these in kamailio 4.4: CC (gcc) [sip-proxy] mem/tlsf_malloc.o mem/tlsf_malloc.c: In function 'tlsf_malloc_init_pkg_manager': mem/tlsf_malloc.c:1353:16: warning: assignment from incompatible pointer type ma.xmalloc = tlsf_malloc; ^ mem/tlsf_malloc.c:1355:16: warning: assignment from incompatible pointer type ma.xrealloc = tlsf_realloc; ^ mem/tlsf_malloc.c:1358:16: warning: assignment from incompatible pointer type ma.xavailable = tlsf_available; ^ mem/tlsf_malloc.c: In function 'tlsf_malloc_init_shm_manager': mem/tlsf_malloc.c:1501:20: warning: assignment from incompatible pointer type ma.xmalloc_unsafe = tlsf_malloc; ^ it would be highly appreciated to get rid of them. -- juha

5 12

calling rpc command in cfg
by Yasin CANER 12 May '16

12 May '16

3 2

speeddial, owners
by Filippo Graziola 11 May '16

11 May '16

Hello everybody, I have a kamailio running on debian, and it works well with a little modification of the kamailio-advanced file. I would like to have some speed dial short numbers for all local users, because I used a random generator script and usernames and password are 25 digits (not only numbers) long. My problem is with the ownership of a every speeddial number. Example: SIP accounts: 5001,5002,5003 SIP domain: mydomain.com add a speeddial number: kamctl speeddial add 5002(a)mydomain.com 51(a)mydomain.com sip:5001@mydomain.com I can call from user 5002 to user 5001 typing 51 correctly. what I would like to do is that sd-id 51(a)mydomain.com could be called from every user registered at mydomain.com. I read the module documentation, kamctl help but I'm not able to find an answer. Could you please help me? Thank you in advance

2 2

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