sr-users September 2015

sr-users@lists.kamailio.org

82 participants
126 discussions

by Igor Potjevlesch

Hello, 3 crashes occurred today (we never seen crashes on this server before) during a period of 20 minutes. I had a look to the coredumps. Here is a quick overview of the first entries of "bt full": core.27671 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27656 : #0 0x00007f557700a048 in free_cell (dead_cell=0x7f5560abbce0) at h_table.c:175 core.27685 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27689 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27691 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.31977 : #0 0x00007f038e66fdee in free_cell (dead_cell=0x7f037811ebe0) at h_table.c:157 core.31994 : #0 0x00007f038e6ab65a in t_forward_nonack (t=0x7f037811ebe0, p_msg=0x7f038fdca438, proxy=0x0, proto=0) at t_fwd.c:1762 core.3700 : #0 0x000000000061697a in fm_insert_free (qm=0x7f19861fa000, frag=0x7f19864d53e8) at mem/f_malloc.c:245 core.3727 : #0 0x00000000006167a4 in fm_extract_free (qm=0x7f19861fa000, frag=0x7f19864f4cc8) at mem/f_malloc.c:208 I don't know how to proceed now. Can you help me to highlight the issue here? Thank you. Regards, Igor.

8 years, 6 months

evapi tcp parsing??

by Jayesh Nambiar

Hello, I'm exploring the evapi module for my kamailio to interface with an external node.js app for third party stuff like AAA, billing engine tasks, notifications and so on. I followed and took some ideas from the rtjson and evapi tutorial found here( http://kb.asipto.com/kamailio:k43-async-sip-routing-nodejs) to build the node.js app consuming events. When I stress tested the scenario using SIPp and tried sending a lot of events at 300-350cps from Kamailio, I noticed that at times the client is receiving 2-3 events in a single message together although I do event_sync_relay once per SIP message received and have netstrings enabled. I believe this is a typical behavior of TCP and needs to be handled by the client using some kind of Netstring handler. Please correct me if I'm wrong. And hence I'd like to know what particularly needs to be taken care of while writing a client that is listening for events on raw tcp socket and how does kamailio handle this situation while receiving messages over TCP socket?? Does kamailio recognize the end of netstring properly on evapi:message-received and give exactly one message to take care of on every "message-received" event or should that be handled in the script somewhere !! I also referred cgrates client over evapi example which is written in GO, but I couldnt find them handling TCP streams clearly either. I'd really appreciate some expert suggestion here to make an informed decision on using the evapi module for a large scale solution. Thanks, - Jayesh

8 years, 6 months

Dialog keep alive options are with error parsing headers

by José Seabra

Hello there, I'm using dialog module to send keep alive options to callee and caller in order to detect if dialog still up, but when kamailio generates the OPTION and parse it, gives the following errors: [root@c4-compute02-srv1 kamailio]# 37(1587) DEBUG: dialog [dlg_hash.c:614]: dlg_lookup(): ref dlg 0x7f89134c1d40 with 1 -> 3 37(1587) DEBUG: dialog [dlg_hash.c:616]: dlg_lookup(): dialog id=2210 found on entry 2350 37(1587) DEBUG: dialog [dlg_req_within.c:427]: dlg_send_ka(): sending OPTIONS to caller 37(1587) DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=< sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [dns_cache.c:3293]: dns_srv_sip_resolve(): (62.28.197.25, 0, 0), ip, ret=0 37(1587) DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 14692 37(1587) DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: 37(1587) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: < sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> 37(1587) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK4693.2af96e95000000000000000000000000.0>; state=16 37(1587) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 37(1587) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 37(1587) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=90d40279ee5745d8851aa989e6fadfc2 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [64]; uri=[sip:unknown@62.28.197.6] 37(1587) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:unknown@62.28.197.6>] 37(1587) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <0> <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 *37(1587) ERROR: <core> [parser/msg_parser.c:96]: get_hdr_field(): ERROR: get_hdr_field: bad header* 37(1587) DEBUG: <core> [parser/msg_parser.c:280]: get_hdr_field(): get_hdr_field: error exit *37(1587) INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [VC* *]* 37(1587) DEBUG: topoh [topoh_mod.c:232]: th_prepare_msg(): parsing headers failed37(1587) DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: 37(1587) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: < sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> 37(1587) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK4693.2af96e95000000000000000000000000.0>; state=16 37(1587) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 37(1587) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 37(1587) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=90d40279ee5745d8851aa989e6fadfc2 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [64]; uri=[sip:unknown@62.28.197.6] 37(1587) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:unknown@62.28.197.6>] 37(1587) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <0> <OPTIONS> 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=e34d43e164ad4b7392dc9c6e1058a983 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: dialog [dlg_cseq.c:215]: dlg_cseq_msg_sent(): traking cseq updates 37(1587) DEBUG: dialog [dlg_hash.c:671]: internal_get_dlg(): ref dlg 0x7f89134c1d40 with 1 -> 4 37(1587) DEBUG: dialog [dlg_hash.c:674]: internal_get_dlg(): dialog callid='13208317017c42128fbf00e1685ee1b3' found on entry 2350, dir=2 37(1587) DEBUG: dialog [dlg_cseq.c:233]: dlg_cseq_msg_sent(): request not going downstream (2) 37(1587) DEBUG: dialog [dlg_hash.c:846]: dlg_unref(): unref dlg 0x7f89134c1d40 with 1 -> 3 37(1587) DEBUG: dialog [dlg_req_within.c:452]: dlg_send_ka(): keep-alive sent to caller 37(1587) DEBUG: dialog [dlg_req_within.c:427]: dlg_send_ka(): sending OPTIONS to callee I'm using the following dialog configuration: modparam("dialog","db_url", DBURL) modparam("dialog","db_mode", 0) modparam("dialog","enable_stats", 1) modparam("dialog","dlg_flag", DLG_FLAG) modparam("dialog","profiles_with_value","concurrentcalls") modparam("dialog", "detect_spirals", 1) modparam("dialog","dlg_match_mode", 2) modparam("dialog", "initial_cbs_inscript", 0) modparam("dialog", "default_timeout", 3600) modparam("dialog", "send_bye", 1) modparam("dialog", "timeout_avp", "$avp(s:timeout_dlg)") modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) I'm also using topoh module. Anyone can help me to identify if I'm doing the proper configuration? Thank you for your support Best Regards José Seabra

8 years, 6 months

Familiar topic about Kamailio with Asterisk behind NAT

by Dirk Teurlings - SIGNET B.V.

Hi, I found some existing topics on this but failed to get a solutions out of them. We're running into some issues with client devices connecting to our private addresses. The way it is setup now: CLIENTS <-> (NAT) <-> INTERNET <-> KAMAILIO(4.2.5) with RTPPROXY(v1) <-> PRIVATE LAN <-> ASTERISK (v1.8) Our Kamaialio and Asterisk are in a private address range, but Kamailio also has a public interface. Most of the clients (about 95%) work well with this setup, but a couple don't. We have one case now where the CLIENT tries to connect to the private address of ASTERISK. And of course, that doesn't work. I'm kind of stuck as to where I need to fix this. I tried using the externaddr option in Asterisk to solve it on that end. But that didn't help anything. The NAT options in Kamailio are not really suited for this, as they tend to fix client NAT problems. Any pointer or help would be greatly appriciated. Cheers, Dirk

8 years, 6 months

Interoperability issues between browsers and mobile phones to place a call.

by Safdar Khan

Hi Daniel sir I have enabled web-sockets module into my kamailio sip server.I am trying to make call from web browsers and mobile phones. To make a call from web browser i am using sipml5 and as a mobile app i am using imsdroid. However i have done some search on net and i found nothing relevant. Any suggestions and help?.Waiting your response eagerly. Thank you

8 years, 6 months

problem with dispatcher load balancing

by Ding Ma

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"); }

8 years, 6 months

Siremis-4.2.0 - error with Alias and login page

by Infinicalls Infinicalls

Hi, I tried installing siremis-4.2.0 following the step given.The Alias was not working. So, I installed it in the /html folder and the installation was completed. But the login page looked strange. There was no logo displayed and the submit button was not there. Mine is apache 4.2 on Ubuntu LTS 14. Kindly let me know how to solve this problem. thanks Ganesh Kumar

8 years, 6 months

FW: Your iptel.org Registration

by Bijal Desai

When I call on sip:bijaldesai@iptel.org, it is not working. Can you please guide me -----Original Message----- From: registrar(a)iptel.org [mailto:registrar@iptel.org] Sent: Tuesday, September 29, 2015 10:00 PM To: Bijal Desai <Bijal(a)ssminfotech.com> Subject: Your iptel.org Registration Thank you for registering with iptel.org. We are reserving the following SIP address for you: sip:bijaldesai@iptel.org To finalize your registration please check the following URL within 24 hours: http://serweb.iptel.org/user/reg/confirmation.php?nr=473f1e2fc74bfb3e9d28eb… (If you confirm later you will have to re-register.) Here are the details of your account: --------------------------------------------------------------------- Username: bijaldesai(a)iptel.org Password: Angel1987@ Email: bijal(a)ssminfotech.com Full Name: Bijal Desai Sip address: sip:bijaldesai@iptel.org (You can always retrieve these via the "Forgot Password" link on the signup page) -- This message is automatically generated by SerWeb.

8 years, 6 months

Kamailio-Asterisk real-time integration with Dynamic IP

by Rudy Eschauzier

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 | | | | |

8 years, 6 months

cnxcc module

by Максим Панчоха

Hello! I'm trying to set max call duration, but after 5 seconds call is continues. Kamailio version: 4.3.2 Here's part of my config file: #!define CC_FLAG 29 ... modparam("cnxcc", "credit_check_period", 1) modparam("cnxcc", "redis", "addr=127.0.0.1;port=6379;db=1") modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) ... if ((is_method("INVITE")) && (is_request()) && (!has_totag())) { route(CNXCC); } route[CNXCC] { setflag(DLG_FLAG); if (!is_method("INVITE")) return; $var(client) = $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$var(client)", "$var(max_time)")) { xlog("Error setting up credit control"); return; } } event_route[dialog:failed] { xlog("L_INFO", "********END CALL*********$DLG_lifetime*******************"); } event_route[dialog:end] { xlog("L_INFO", "******END CALL************$DLG_lifetime****************"); } event_route[dialog:start] { xlog("L_INFO", "***********************DIALOG STARTED****************************"); } event_route[cnxcc:call-shutdown] { $var(duration) = $TS - $dlg(start_ts); xlog("L_INFO", *****END CALL***BALANCE IS EMPTY************$var(duration)*************"); } Tell me please, where I made a mistake. Thank you for a response.

8 years, 6 months

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sr-users September 2015