sr-users June 2015

sr-users@lists.kamailio.org

82 participants
136 discussions

Kamailio TLS
by Kethzer Docteur 12 Feb '16

12 Feb '16

Im trying to enable Kamailio TLS but the configuration is not running at all when I cahnge it liike this as follow. Kamailio 3.3 ubuntu 12.04 LTS #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif Please help -- Kethzer Docteur

4 3

Append to reply in default action of failure_route
by Efelin Novak 28 Dec '15

28 Dec '15

Hi, I would like to append a header to a 'winning' negative reply in failure_route and let the Kamailio do the default action (state fully forward the winning reply). When I use append_to_reply("Foo: bar\r\n"); and then call exit; in failure_route nothing is appended. When I use same append_to_reply then t_relay("505","Error"); and exit; the header is appended. When I use append and t_reply with dialog modul turned on I got a bug I'm solving here '[SR-Users] t_reply in failure route with dialog module'. So my question is how to put a header into a reply when I don't want to alter its code or text? I'm using Kamilio 4.0.4 on Debian 7.1 Thanks for an answer Efelin

3 3

kamailio 4.0.1 websocket configuration
by Rupayan Dutta 28 Dec '15

28 Dec '15

Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta

4 3

Kamailio - Asterisk, problem with SIP trunk mode
by Mickael MONSIEUR 02 Dec '15

02 Dec '15

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

3 6

Issue with SNMPstats
by Igor Potjevlesch 24 Nov '15

24 Nov '15

Hello, I have configured the SNMPstats module fine in my machine because I get the SNMP information with the new KAMAILIO MIBS but I don't understand why the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 However I can see the transactions: KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryTotalTransactions.0 = Counter32: 54797 Should I configure an specific parametr of the SNMPstats module? Regards,

3 6

Kamailio IMS - VM
by Bruno Cruz 17 Nov '15

17 Nov '15

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

4 8

Header field Contact not changed on kamailio 4.2
by Зуев Михаил 09 Nov '15

09 Nov '15

Hi all, I need to modify the host part of a contact header. I'm trying something like: route { ..... if (remove_hf("Contact")){ if (insert_hf("Contact: rn", "Contact")) xlog("Contact modified"); } if (!t_relay("YYY.YYY.YYY.YYY","5060")) { sl_reply_error(); } ...... } When i look SIP DUMP on YYY.YYY.YYY.YYY side i see older Contact field. I try subst_hf with regexp, subst and simply remove Contact, all this does not work. What i am doing wrong? -- С уважением Зуев Михаил

3 2

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"); }

2 10

A question to you, SIP providers: what is your policy on PBXes calling themselves
by Antonio Gómez Soto 23 Aug '15

23 Aug '15

Hello, This is question on PBX behavior, what is the right thing to do, and how do PBX's generally behave. If a user on a phone, dials a number, which happens to be configured on the same phone system (for example another tenant), there are two options: 1. The PBX notices this, and directly connects the phone to the DID on that system (breaking separation of tenants) 2. The PBX sends the call out on the SIP trunk, and the provider-routing sends the call back as an incoming call. What are the pros and cons of each option from the SIP provider point of view? How do PBX's generally behave? Thanks, Antonio PS: I reposted, because the original question was apparently not phrased in a clear way.

3 2

IPSec supporting open source SIP Server
by Priyaranjan Nayak 23 Aug '15

23 Aug '15

Hi All, I need one IPSec supported opensource SIP server for testing my sip client endpoint.Could you please tell me which SIP server supporting IPSec ? Thanks in advance. Thanks Priyaranjan

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sr-users June 2015