sr-users August 2015

sr-users@lists.kamailio.org

79 participants
124 discussions

problem with dispatcher load balancing
by Ding Ma 05 Oct '15

05 Oct '15

I'm trying to set up kamailio dispatcher to distribute calls to 2 asterisk servers. So far, the failover case seems ok, but I cannot get the dispatcher to distribute load. All calls are going to the last destination entry in the dispatcher table even if I have set the maxload attributes. I'm using algorithm 10 for load distribution. The number of calls sent to one asterisk is well above the maxload. Hope someone more experienced in dispatcher can review my config and give me some recommendations. Thanks in advance. Here is the content of dispatcher table in postgresql db. select * from dispatcher; id | setid | destination | flags | priority | attrs | description ----+-------+--------------------+-------+----------+---------------------------+------------- 1 | 1 | sip:10.0.1.31:5061 | 0 | 0 | duid=asterisk1;maxload=25 | Asterisk1 2 | 1 | sip:10.0.1.33:5061 | 0 | 0 | duid=asterisk2;maxload=25 | Asterisk2 (2 rows) Here are the dispatcher sections in kamailio.cfg. # ----- dispatcher params ----- #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", DBASTURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "force_dst", 1). # If flag 2 is set, then failover support is enabled. modparam("dispatcher", "flags", 3) # the last address in destination set is used as a final option to send the request to modparam("dispatcher", "use_default", 1) # load balancing fail over modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # PVs for hashing modparam("dispatcher", "hash_pvar", "$fU@$ci") # PVs to store results when calling ds_is_from_list modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") # method to probe the gateways modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:dispatcher@localhost") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) # size of hash table storing data for call load dispatching, power of two modparam("dispatcher", "ds_hash_size", 10) # expiration time in seconds to remove the load on a destination if no BYE was received modparam("dispatcher", "ds_hash_expire", 3600) # expiration time in seconds to remove the load on a destination if no 200 OK for INVITE was received # and state updated with ds_load_update modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 30) #!endif route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_DISPATCHER if(is_method("BYE|CANCEL") && ds_is_from_list("1", "3")) ds_load_update(); #!endif ...... } } route[FROMASTERISK] { #!ifdef WITH_DISPATCHER if(ds_is_from_list("1", "3")) { xlog("L_DBG","$rm from $fU@$si:$sp: Call from Asterisk cluster\n"); return 1; } return -1; #!else if ($si==$sel(cfg_get.asterisk.bindip)) { return 1; } return -1; #!endif } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); #!ifdef WITH_DISPATCHER if(is_method("INVITE") && ds_is_from_list("1", "3")) { if(status=~"2[0-9][0-9]") { ds_load_update(); } else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]") route(NATMANAGE); } #!ifdef WITH_DISPATCHER failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { # mark the destination Inactive and Probing ds_mark_dst("IP"); # select the new destination if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } else { # last available node failed to reply, no other destinations available send_reply("404", "No destination"); exit; } } } #!endif route[TOASTERISK] { #!ifdef WITH_DISPATCHER # ds_mark_dst("IP"); # Call load distribution if(!ds_select_dst("1", "10")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","$rm from $fU@$si:$sp: No destinations available for $rd\n"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); #!else $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); #!endif route(RELAY); exit; } # Dispatcher detects a destination goes down event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du)\n"); } # Dispatcher detects a destination comes up event_route[dispatcher:dst-up] { xlog("L_ERR", "Destination up: $rm $ru\n"); }

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Kamailio-Asterisk real-time integration with Dynamic IP
by Rudy Eschauzier 02 Oct '15

02 Oct '15

All, The Kamailio-Asterisk integration that is described here asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [Asipto - SIP and VoIP Knowledge Base Site] is limited to Asterisk servers with a fixed IP address. The reason is that Kamailio identifies and authorizes traffic from Asterisk based on the source IP address. Attached is a patch to kamailio.cfg that adds support for dynamic Asterisk hosts. It introduces the option to define a fully-qualified domain name instead of the IP address to identify the Asterisk server. The host name can be dynamically updated to always point at the (changing) IP address of the Asterisk server. Defining WITH_ASTERISK_FQDN enables the new option. When WITH_ASTERISK_FQDN is set, Kamailio will perform a DNS search on the host name in the asterisk.bindhost variable. It will authorize incoming traffic if it originates from one of the IP addresses the DNS lookup returns (and matches the asterisk.bindport variable as before). Furthermore, the patch introduce the option to automatically set the IP address of the server that Kamailio is running on. This does away with the need to properly define kamailio.bindip on every server that the configuration is running on. This option is enabled by defining WITH_AUTOIP. With the suggested changes I have been able to successfully run an Asterisk server in my home, having a dynamic public IP address and connecting to the Kamailio proxy on a VPS server with a static IP addres. Please let me know if you have any comments or suggestions. Kind regards,Rudy Eschauzier. | | | | | | | | | | | asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [...Table of Contents Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database Architecture Registration Call Initiation Requirements MySQL Insta... | | | | View on kb.asipto.com | Preview by Yahoo | | | | |

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Proxy-Require: privacy error
by Chad 21 Sep '15

21 Sep '15

When a call comes in with a SIP header: "Proxy-Require: privacy" line kamailio throws this error: Aug 21 14:27:57 load1 /usr/sbin/kamailio[29447]: ERROR: <core> [parser/hf.c:224]: dump_hdr_field(): DEBUG: dump_hdr_field: type=18, name=Proxy-Require, body=privacy, parsed=(nil), next=0x7efc42f15150 Aug 21 14:27:57 load1 /usr/sbin/kamailio[29446]: ERROR: <core> [parser/hf.c:224]: dump_hdr_field(): DEBUG: dump_hdr_field: type=18, name=Proxy-Require, body=privacy, parsed=(nil), next=0x7efc42f0a7b0 Aug 21 14:27:58 load1 /usr/sbin/kamailio[29447]: ERROR: <core> [parser/hf.c:224]: dump_hdr_field(): DEBUG: dump_hdr_field: type=18, name=Proxy-Require, body=privacy, parsed=(nil), next=0x7efc42f13180 Aug 21 14:27:58 load1 /usr/sbin/kamailio[29446]: ERROR: <core> [parser/hf.c:224]: dump_hdr_field(): DEBUG: dump_hdr_field: type=18, name=Proxy-Require, body=privacy, parsed=(nil), next=0x7efc42f0dfb8 Anyone know why that happens or how to fix it? -- ^C Chad

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Using PSTN as "fallback"
by Michael Nielsen 09 Sep '15

09 Sep '15

I have Kamailio running and connected to a PSTN gateway. My subscribers are named ex. +442071234567 - same as their real GSM number from my PSTN gateway. I'm using the standard kamailio.cfg which ships with version 4.3. When I'm trying to dial SIP client to SIP client I would like to have Kamailio route the call internally if a subscriber exists with ex. +442071234567. If no subscriber exists with ex. +442071234567 it should send it to my PSTN gateway. As it is now it seems as if it are trying to both call "internally" and via the PSTN gateway. How should one fix this issue the best way?

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core parameter onsend_route_reply=1 doesn't execute onsend_route block for received replies
by Julia Boudniatsky 07 Sep '15

07 Sep '15

Hello, I set some logs in onsend_route and it works fine for requests. With setting core parameter onsend_route_reply=1 ( yes, on), i expect to see logs for replays also, but it not occurs. Is this a bug or I miss something? version: kamailio 4.2.6 (x86_64/linux) db77ac BR, Julia

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Siremis 4.2 with PHP 5.6
by Patrick Bezensiertblubb 07 Sep '15

07 Sep '15

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Re: [SR-Users] CRITICAL log on kamailio startup
by Nuno Reis 04 Sep '15

04 Sep '15

Good morning. Has anyone seen this too? Looking forward to hear from you. BR, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/> On Fri, Aug 14, 2015 at 1:41 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: > Hi guys. > > I've noticed this CRITICAL log line on kamailio startup: > > CRITICAL: dialog [dlg_cb.c:137]: register_dlgcb(): DLGCB_CREATED type > registered after shutdown!?! > > > This line started to show up after upgrade from 4.2.x to 4.3 and now to > 4.3.1. > Everything seems to be working fine and a really don't see any harm > anywhere. > Could you please comment on this? > > Cheers, > > -- > > *Nuno Miguel Reis* | *Unified Communication** Systems* > M. +351 913907481 | nreis(a)wavecom.pt > WAVECOM-Soluções Rádio, S.A. > Cacia Park | Rua do Progresso, Lote 15 > 3800-639 AVEIRO | Portugal > T. +351 309 700 225 | F. +351 234 919 191 > *GPS > <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> > | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* > > <http://www.wavecom.pt/> > <http://www.wavecom.pt/pt/> > > >

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Kamailio memmory leak after it does not receive response for CANCEL message
by Bui Minh Yen 03 Sep '15

03 Sep '15

Hi everyone, I am using Kamailio-4.1.5 and Kamailio-4.2.5 but with this both version I had a problem: Connection diagrams: SIP client ---- SBC --- Kamailio --- MGC/MGW --- GSM/PSTN subscriber Scenario: - Caller (SIP) make a new call to GSM (PSTN) subscriber - When GSM/PSTN is ringing or not ringing, Caller cancel call (Caller send CANCEL message) - But MGC does not send response message for CANCEL message - After about 120s (fr_inv_timer), Kamailio send 487 Request Terminated to SBC. >From this time, Kamailio have memory leak (tm does not free transactions) I reproduced this bugs 4-5 times and same problem is happen I attached capture file and log when I show shared memory (using kamcmd core.shmmem) and tm statistics (using kamcmd tm.stats). And time Kamailio start memory leak is 08-30-15-10:18:53 Could you help me solve this problem? Thank you so much! -- --------------------------- *Bui Thi Minh Yen*

2 2

during registration nonce expired, after backwards time shift
by Vasiliy Ganchev 03 Sep '15

03 Sep '15

Hi list! (sorry for sending first mail to wrong sub forum) Have a following issue: Server with Kamailio restart with wrong time (why - this is separate part of my investigations), timeshift e.g. for 2 hours in feature. After ntp daemon adjust correct time (move time backwards), Kamailio do not accept REGISTERs, pv_www_authenticate answer with code -4 (nonce expired) I've reade the description of: http://kamailio.org/docs/modules/4.2.x/modules/auth.html#auth.p.nonce_expire and http://kamailio.org/docs/modules/4.2.x/modules/auth.html#auth.p.nonce_auth_… This descriptions mention situation with backwards timeshift from future. As I understand, Kamailio do not authorise the registration, whose nonces where generated in future (before ntp shifted the time). But in my tests, Kamailio do not accept even REGISTER from UA, sent after time shift (with new nonces, that as I expected has to use correct time for nonce generation, and be authorised with no problem). Only Kamailio process restart help to cope with the issue. It looks like for nonce generating Kamailio use old time (in feature, that was before timeshift). I wander, does the Kamailio behave in described case - as is expected? If I am wrong in some of my assumptions, please point out. version: kamailio 4.2.5 Thank in advance! Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/during-registration-nonce-expired-a… Sent from the Users mailing list archive at Nabble.com.

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Kamailio TLS handshake fails above 6500 TLS connections
by Priyaranjan Nayak 02 Sep '15

02 Sep '15

Hi All, We are trying to make 10000 tls connection with Kamailio server for SIP signaling.When I try beyond 6500 TLS connection then TLS handshake is failing.We are running the kamailio by using below command - kamailio -f kamailio.cfg -N 40 -m 1024 -M 16 NOTE: In kamailio.cfg, we have mentioned tls_max_connections=60000 Could you please guide me, how can I improve the tls connections with kamailio ? Thanks Priyaranjan

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sr-users August 2015