sr-users June 2013

sr-users@lists.kamailio.org

94 participants
171 discussions

cnxcc developer module pulse concept
by Eduardo Lejarreta 06 Jun '13

06 Jun '13

Good morning Carlos. On cnxcc_set_max_credit() function I don't understand the concept "pulse" (sure it's something easy but I'm a not aware about this matter) Could you please explain it in an example? Thanks and best regards. -- Eduardo Lejarreta.

1 0

if (t_check_status("486|408"))
by hiro 06 Jun '13

06 Jun '13

i'm trying to use the example kamailio.cfg to route to voicemail server on busy or decline. Only thing I did was adding decline code to t_check_status("486|408"), enabling the preprocessor variable for voicemail and changing the voicemail host and port to my voicemail server. No requests arrive on my voicemail server and the dial tone keeps ringing even if phone is busy and when I decline on the receiving phone there's a new INVITE sent to the phone directly afterwards. Am I doing something wrong here?

3 15

Msilo dumping offline messages
by sipatse＠in.gr 05 Jun '13

05 Jun '13

2 1

dialog ka-timer/ka_interval results in flood op OPTIONS
by Daniel Tryba 05 Jun '13

05 Jun '13

Trying to use the dialog module to implement calllimits I'm running into a problem when trying to use keep-alives to detect dead dialogs. Limits work fine without keepalives. But as soon as either dlg_set_property("ka-dst"); or dlg_set_property("ka-src"); are used, after the ka_interval timeout kamailio will start to send OPTIONS as fast as possible. There is no difference in behavior for inbound of outbound calls. Older threads on this seem to end without solution. Increasing shmem/pkgmem makes no difference. System is running on Debian/stable with deb http://deb.kamailio.org/kamailio wheezy main packages. # kamailio -V version: kamailio 4.0.0 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 13:13:24 Mar 29 2013 with gcc 4.7.2 modparam("dialog","dlg_flag",4) modparam("dialog","hash_size",4096) modparam("dialog","enable_stats",1) modparam("dialog","db_url",DBURL) modparam("dialog","db_mode", 2) modparam("dialog","db_update_period", 15) modparam("dialog","profiles_with_value","maxchannels") modparam("dialog","profiles_no_value", "all") modparam("dialog","dlg_match_mode", 1) modparam("dialog","send_bye", 1) modparam("dialog","default_timeout", 43200) modparam("dialog","ka_timer", 15) modparam("dialog","ka_interval", 30) dlm_manage() uses as: route[DIALOG] { if (is_method("CANCEL|ACK") || (has_totag() && is_method("INVITE| BYE"))) { dlg_manage(); return; } if (is_method("INVITE") && !has_totag()) { if($au==$null) { route(MAXCALLSINBOUND); } else { route(MAXCALLSOUTBOUND); } dlg_manage(); } } route[MAXCALLSOUTBOUND] { if(method=="INVITE") { if($au && $avp(src_maxchannels)>0) { $var(maxchannels) = 0; get_profile_size("maxchannels", "$au", "$var(maxchannels)"); if($var(maxchannels)>=$avp(src_maxchannels)) { send_reply("403","Call limit reached"); exit; } set_dlg_profile("maxchannels","$au"); dlg_set_property("ka-src"); get_profile_size("maxchannels", "$au", "$var(maxchannels)"); } } return; } MAXCALLSINBOUND is nearly the same but uses $rU instead of $au (and routes to voicemail/redirects if applicable). Resulting in network traffic: U +6.719318 10.0.32.42:5060 -> 10.0.34.226:27819 OPTIONS sip:+31880100781@10.0.34.226:27819 SIP/2.0. Via: SIP/2.0/UDP 10.0.32.42;branch=z9hG4bK3548.6baca897.0. To: sip:+31880100781@10.0.32.42;tag=3c64a586e73b0dc9o0. From: sip:0880100705@10.0.32.42;tag=as2b14090d. CSeq: 101 OPTIONS. Call-ID: db688a0a-53e2d315@localhost. Max-Forwards: 70. Content-Length: 0. . # U +0.000041 10.0.32.42:5060 -> 10.0.34.226:27819 OPTIONS sip:+31880100781@10.0.34.226:27819 SIP/2.0. Via: SIP/2.0/UDP 10.0.32.42;branch=z9hG4bK3548.7baca897.0. To: sip:+31880100781@10.0.32.42;tag=3c64a586e73b0dc9o0. From: sip:0880100705@10.0.32.42;tag=as2b14090d. CSeq: 101 OPTIONS. Call-ID: db688a0a-53e2d315@localhost. Max-Forwards: 70. Content-Length: 0. . # U +0.000028 10.0.32.42:5060 -> 10.0.34.226:27819 OPTIONS sip:+31880100781@10.0.34.226:27819 SIP/2.0. Via: SIP/2.0/UDP 10.0.32.42;branch=z9hG4bK3548.8baca897.0. To: sip:+31880100781@10.0.32.42;tag=3c64a586e73b0dc9o0. From: sip:0880100705@10.0.32.42;tag=as2b14090d. CSeq: 101 OPTIONS. Call-ID: db688a0a-53e2d315@localhost. Max-Forwards: 70. Content-Length: 0. . # U +0.000026 10.0.32.42:5060 -> 10.0.34.226:27819 OPTIONS sip:+31880100781@10.0.34.226:27819 SIP/2.0. Via: SIP/2.0/UDP 10.0.32.42;branch=z9hG4bK3548.9baca897.0. To: sip:+31880100781@10.0.32.42;tag=3c64a586e73b0dc9o0. From: sip:0880100705@10.0.32.42;tag=as2b14090d. CSeq: 101 OPTIONS. Call-ID: db688a0a-53e2d315@localhost. Max-Forwards: 70. Content-Length: 0. . [6412 more OPTIONS before hanging up or phone reboots] syslog output (another call since turning debug on will choke the machine in disk IO to syslog), it keeps repeating stuff like this: 14:19:56 DEBUG: topoh [topoh_mod.c:345]: the COOKIE is [xxx] 14:19:56 DEBUG: dialog [dlg_req_within.c:387]: keep-alive sent to caller 14:19:56 DEBUG: dialog [dlg_hash.c:753]: unref dlg 0x7fb4bfe3d2e0 with 1 -> 2 14:19:56 DEBUG: dialog [dlg_hash.c:588]: ref dlg 0x7fb4bfe3d2e0 with 1 -> 3 14:19:56 DEBUG: dialog [dlg_hash.c:590]: dialog id=7871 found on entry 2710 14:19:56 DEBUG: dialog [dlg_req_within.c:366]: sending BYE to caller 14:19:56 DEBUG: tm [uac.c:243]: DEBUG:tm:t_uac: next_hop=<sip: +31880100781@10.0.34.226:27819> 14:19:56 DEBUG: tm [uac.c:182]: DEBUG: dlg2hash: 6504 14:19:56 DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: 14:19:56 DEBUG: <core> [parser/msg_parser.c:625]: method: <OPTIONS> 14:19:56 DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip: +31880100781@10.0.34.226:27819> 14:19:56 DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> 14:19:56 DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK8691.df660fd1.0>; state=16 14:19:56 DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 14:19:56 DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 14:19:56 DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 14:19:56 DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=a2b76e7ff411361o0 14:19:56 DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 14:19:56 DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [54]; uri=[sip:+31880100781@10.0.32.42] 14:19:56 DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [sip: +31880100781@109. 235.32.42] 14:19:56 DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <101> <OPTIONS> 14:19:56 DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 14:19:56 DEBUG: <core> [parser/msg_parser.c:106]: found end of header 14:19:56 DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=as23ba5e76 14:19:56 DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 3

Re: [SR-Users] kamcmd vs kamctl
by DanB 05 Jun '13

05 Jun '13

Hey Ovidiu, Thanks for the tip. Will have a look. Have a good one! DanB

1 0

Asterisk 11.4.0+Kamailio 4.0.1 Integration
by Ravindra Gowda 04 Jun '13

04 Jun '13

Dear All, I am trying to integrate kamailio 4.0.1 server with Asterisk 11.4.0 in Ubuntu 12.04 LTS version. I tried it by using the following link : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. But after all, when i tried to start Kmailio server with /etc/init.d/kamailio start command , i am getting the following errors: asterisk-kamailio@asterisk- kamailio:/$ sudo /etc/init.d/kamailio start [sudo] password for asterisk-kamailio: /etc/init.d/kamailio: 1: /etc/init.d/kamailio: sed: not found /etc/init.d/kamailio: 118: /etc/init.d/kamailio: arithmetic expression: expecting primary: "" what could be the wrong, and how can i solve it? Please share your valuable suggestions on this. Any help will greatly appreciate. *Thank you and Best regards,* *R Ravindra **Gowda* *Engineer - Systems and Products*. *Thrikasa Technologies* *| Office: +91 40 23260434 | Mob: +918885284050 * *| Fax:+91 40 23221045* *| email: ravindra(a)thrikasa.in* *| Web: www.thrikasa.in* 'If you don't give up,you cannot fail'. The information contained in this communication is proprietary to Thrikasa Technologies., and/or third parties, may contain classified or privileged information, and is intended only for the use of the intended addressee thereof. If you are not the intended addressee, please be aware that any use, disclosure, distribution and/or copying of this communication is strictly prohibited.If you receive this communication in error, please notify the sender immediately and delete it from your computer.

2 1

Re: [SR-Users] Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1
by Michael Leuker 04 Jun '13

04 Jun '13

Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described: Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy: chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" <sip:xxxxxxxx@198.23.139.21>;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy I doubt that I can make the necessary modifications even with your hints, but would be willing to pay for your (or anybody's) time solving the matter in a way that (a) leaves the FreePBX installation untouched as far as possible and (b) is easy to apply to subsequent versions. Ideally, it would be a slight modification of Daniel's excellent tutorial at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb, allowing to create extensions in FreePBX. Anybody willing to give it a shot? Just let me know what you would charge and we can talk about it.

2 5

Re: [SR-Users] kamcmd vs kamctl
by DanB 04 Jun '13

04 Jun '13

Hey Daniel, Thanks for the answer. It makes more sense now. Have a good one! DanB

2 1

Re: [SR-Users] [Spce-user] FYI: Package update(s) in release 2.8
by Andrew Pogrebennyk 04 Jun '13

04 Jun '13

Dear 2.8 users, this update includes the fixes for kamailio-lb config (multiple sockets-related): - added record-route param to store socket; review logic for relaying in-dialog requests (fallback to default send socket if it's not defined explicitly); fixed socket selection for ACK after 4xx and replies to in-dialog requests to callee; - also file descriptors limit has been lifted for sems. The upgrade is highly recommended if you run the 2.8 version. On 06/04/2013 12:16 PM, Sipwise Repository Tracker wrote: > One or more updates have been released in NGCP release 2.8: > > * ngcp-templates-ce-asterisk was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-cdr-exporter was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-check-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-cleanup-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-hylafaxplus was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-hylafaxplus-diva was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-hylafaxplus-iax was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-iaxmodem was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-kamailio was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-lsb was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-mediaproxy-ng was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-mediator was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-mysql was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-nginx was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-odbc was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-reminder was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-sems was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-system was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-ce-vmnotify was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-asterisk was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-callingcard was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-cdr-exporter was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-check-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-cleanup-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-glusterfs was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-ha was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-heartbeat2 was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-hylafaxplus was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-hylafaxplus-diva was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-hylafaxplus-iax was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-iaxmodem was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-kamailio was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-lsb was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-mediaproxy-ng was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-mediator was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-monit was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-monitoring-tools was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-mysql was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-nginx was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-odbc was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-pushd was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-redis was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-reminder was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-sems was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-system was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-vmnotify was updated from version 2.8.6 to version 2.8.7 > * ngcp-templates-pro-voisniff-ng was updated from version 2.8.6 to version 2.8.7 > > To apply the latest update(s) execute: > > apt-get update && apt-get upgrade && \ > ngcp-update-db-schema && ngcp-update-cfg-schema && \ > ngcpcfg apply > > If you encounter any problems please report a bug at > the spce-user(a)lists.sipwise.com mailing list. > > Further documentation is available at: > > http://www.sipwise.com/products/spce/documentation/ > > This is an automated mail by the Continuous Delivery platform at Sipwise.com > > _______________________________________________ > Spce-user mailing list > Spce-user(a)lists.sipwise.com > http://lists.sipwise.com/listinfo/spce-user >

2 1

Kamailio and Siremis
by Isaac A. McDonald 04 Jun '13

04 Jun '13

Hello All, I'm a bit confused as to how Siremis interacts with Kamailio. From what I understand, the dial-plan is done in the kamailio.cfg with regex statements and requires a restart of the Kamailio process to take effect. How then does Siremis change the dial-plan in Kamailio without writing re-writing the config? I'm assuming Siremis is merely writing to a MySQL database and some additional code is required in the kamailio.cfg in order to be able to use the dial-plan functionality of Siremis? What must I change in the kamailio.cfg to be able to use Siremis for dial-plan modification?

2 1

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sr-users June 2013