sr-users February 2012

sr-users@lists.kamailio.org

79 participants
139 discussions

Have-my-domain! - host your own SIP domain at iptel.org
by SuperCel.mx.tc 26 Feb '12

26 Feb '12

Sirs: We cannot get our domain to configure to the iptel SIP Service. I have attached a copy of how we've tried setting the DNS, but we cannot find anywhere that gives us info on the "SIP" and/or "protocol UDP". Can you assist us so that we can use our own domain (http://supercel.tel) so that we can send SIP calls across this domain? Sincerely, Frank Riley, CTO SuperCel.tel

1 0

Kamailio segmentation fault
by Fabian Borot 24 Feb '12

24 Feb '12

HelloOur kamailio proxy (version 1.5.2 ) crashed today due to a segmentation fault:Feb 22 16:52:10 AW-PROXY-01 kernel: kamailio[3667]: segfault at 000000000000000b rip 00002ab2b47bf93f rsp 00007ffff62dd270 error 6Apparently due to lack of private memory:[root@AW-PROXY-01 log]# cat proxy.log | grep ERROR | grep dbFeb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_rows: no private memory left Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_result: converting rows failed Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_store_result: failed to convert result Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:core:db_do_query: error while storing result Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_fetch_result: no private memory left Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:htable:ht_db_load_table: Error while fetching result I was hoping to see a core file somewhere but I haven't found one yet. I did not have the line "disable_core_dump=no" set on the config file because supposedly is set to no by default: >From the online guide for my version: //////////////////// Disable_core_dumpCan be 'yes' or 'no'. By default core dump limits are set to unlimited or a high enough value. Set this config variable to 'yes' to disable core dump-ing (will set core limits to 0).Default value is 'no'. ///////////////////How can I make sure that a core file is created after a situation like this? Where will it be created? How can I check the status of the private memory? I use the "kamctl fifo get_statistics all" but I don't know which one relates to "private memory": Example:shmem:total_size = 2147483648 shmem:used_size = 16860928 shmem:real_used_size = 17033968 shmem:max_used_size = 19174528 shmem:free_size = 2130449680 shmem:fragments = 1850I know that with so little information is hard to diagnose, but if anybody has an idea as to what else I can do I greatly appreciate it. Thank you in advancefborot

3 7

avp_delete problem [kamailio 1.5]
by Sven Knoblich 24 Feb '12

24 Feb '12

Hello all, i am currently confused by using the avpops function avp_delete. When i run avp_delete without the flag-value \g only the last value will be unset (like an undo). Is this the wanted behaviour? EXAMPLE: $avp(test) = "test" if( $avp(test)){xlog("L_NOTICE","result1:$avp(test)\n");} $avp(test) = "nooo"; if( $avp(test)){xlog("L_NOTICE","result2:$avp(test)\n");} avp_delete("$avp(test)"); if( $avp(test)){xlog("L_NOTICE","result3:$avp(test)\n");} RESULT: result1:test result2:nooo result3:test Could anybody helps me to understand that? thanks in advance, Sven

4 5

Adding parameter to headers
by Uri Shacked 24 Feb '12

24 Feb '12

Hi, I need to add a parameter to the contact header. suppose to look like: Contact: <sip:anonymous@domain:port;param1=XXX;param2=YYY> How do I add the parameters? Should i concatenate the $ct with the parameters? Or is there a better way? BR, Uri

2 1

fix_nated_sdp issue
by Ric Marques 24 Feb '12

24 Feb '12

Greetings, I'm not sure if I found a bug, or if I just have something completely misconfigured... I'm a total newb with Kamailio, working on a proof of concept design. Here's my configuration: provider -> nat firewall -> kamailio/rtpproxy -> asterisk For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use fix_nated_sdp("2", "10.50.50.8") to rewrite the media ip address to resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What I'm running into is the 'c=' line doesn't get re-written properly... it inserts the specified address in front of the existing address, and I end up with the following line in my INVITE: c=IN IP4 10.50.50.810.0.10.10 I have the fix_nated_sdp command under route[sipout], because I only want to use it on calls being sent outside the nat firewall. Here's the sip invite without the 'fix_nated_sdp' command: -------------------------------------------------------------------------------------------------------------- INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes> Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060 Max-Forwards: 69 From: "1009" <sip:1009@10.0.10.11>;tag=as5498b77e To: <sip:19165551212@xxx.xxx.xxx.xxx> Contact: <sip:1009@10.0.10.11:5060> Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:06:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 309 P-hint: outbound v=0 o=root 604360056 604360056 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.0.10.10 t=0 0 m=audio 9702 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------- Here's the sip invite with the 'fix_nated_sdp' command: -------------------------------------------------------------------------------------------------------------- INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes> Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060 Max-Forwards: 69 From: "1009" <sip:1009@10.0.10.11>;tag=as49e00c81 To: <sip:19167828326@xxx.xxx.xxx.xxx> Contact: <sip:1009@10.0.10.11:5060> Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:18:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 347 P-hint: outbound v=0 o=root 1009117068 1009117068 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.50.50.8.10.0.10.10 t=0 0 m=audio 13540 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=oldmediaip:10.0.10.11 a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------- Is this a bug, or is it likely I have something else screwed up? Thank you in advance for your assistance - this list is an incredible resource! -Ric

2 3

Require Certificate
by Bruno Bresciani 22 Feb '12

22 Feb '12

Hi All, Does it possible on tls module require certificates only some hosts? My kamailio comunicate with several UA clients (Softphones and IP phones) and one SIP gateway. I want require certificate only gateway SIP, it's possible? The parameter *modparam("tls", "require_certificate", 1)* require certificate of all clients... Best Regards

2 4

Topoh hiding prefixes
by Efelin Novak 22 Feb '12

22 Feb '12

Hi folks, I have a topology like this: UA -> kamailio -> PSTN GW. When I'm placing calls to a PSTN I have to append prefixes so the PSTN GW knows how to route the call. So the request-uri is <prefix><number>@<gateway_ip> e.g. 999123456(a)192.168.1.1. When the UA calls to the GW a "200 OK" message contains a contact header with 999123456(a)192.168.1.1. This is a problem because I don't want my customers to see gateway prefixes. The Topoh module only hides IP addresses or the domain part. Is there any way to remove prefixes. This document (http://www.kamailio.org/events/2011-Cluecon/DCM-kamailio-security.pdf) stated "encoding IP and prefixes can be set via parameters". Can I remove prefixes from the contact in any other way than some contact header substitutions? Thanks Efelin

3 9

CallControl and MediaProxy
by Reda Aouad 22 Feb '12

22 Feb '12

Hi, I noticed that calling call_control( ) function forces engage_media_proxy( ). Is there a way to disable this behavior ? Thanks Reda

1 0

Kamailio v3.2 problem with dialog module.
by Phillman25 Kyriacou 21 Feb '12

21 Feb '12

Dear List After upgrading from V3.1.5 to V3.2. Kamailio crashes with the below error messages: Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [db.c:392]: invalid number of rows received: 4, dialog_vars Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [db.c:419]: querying version for table dialog_vars Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: dialog [dlg_db_handler.c:153]: error during dialog-vars version check. Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: dialog [dialog.c:646]: failed to initialize the DB support Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) My dialog paramaters are as follows: # ----------------- DIALOG MODULE PARAMETERS----------------------------------# modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "default_timeout", 21600) modparam("dialog", "dlg_extra_hdrs", "NULL") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 60) modparam("dialog", "db_fetch_rows", 500) modparam("dialog", "table_name", "dialog") modparam("dialog", "from_uri_column", "from_uri") modparam("dialog", "from_tag_column", "from_tag") modparam("dialog", "to_uri_column", "to_uri") modparam("dialog", "to_tag_column", "to_tag") modparam("dialog", "h_id_column", "hash_id") modparam("dialog", "h_entry_column", "hash_entry") modparam("dialog", "state_column", "state") modparam("dialog", "start_time_column", "start_time") modparam("dialog", "timeout_column", "timeout") modparam("dialog", "sflags_column", "sflags") modparam("dialog", "bridge_controller", "sip:controller@kamailio.org") modparam("dialog", "default_timeout", 7200) i realised that when i set the "modparam("dialog", "db_mode", 0)" kamailio starts fine, seems to be a problem connecting to the database. Does anyone know what is wrong? Thanking you in advance Phillip

2 3

UAC canceling a specific early dialog with BYE
by Andrew Pogrebennyk 21 Feb '12

21 Feb '12

Hello, we have a Kamailio proxy which gets the call from PSTN gw and does some call forwarding (serial forking) to several destinations through our sbc The call flow I am looking at is: - Kamailio sends INVITE to branch_1. - branch_1 sends 180 with to-tag*, proxy relays it to the gw * 180 meets the requirements for dialog creating 18x responses in sections 12.1, 12.1.1 because it contains to-tag, contact and mirrored record-route. - After some seconds Kamailio sends a CANCEL to branch_1. - And sends the INVITE to branch_2. - branch_1 replies 200 for the CANCEL and 487 for the INVITE. - branch_2 replies 180 and 200 for the INVITE. - When PSTN gw receives that it sees it still needs to cancel other early dialog established by 180 from branch_1. - The PSTN gw sends a BYE with to-tag of branch_1 to cancel this specific early dialog. SIP allows early dialogs to individually released while other dialogs continue, as written in RFC, section 15: "The BYE request is used to terminate a specific session or attempted session. In this case, the specific session is the one with the peer UA on the other side of the dialog. (...). The caller’s UA MAY send a BYE for either confirmed or early dialogs, and the callee’s UA MAY send a BYE on confirmed dialogs, but MUST NOT send a BYE on early dialogs." The BYE follows the loose routing path, proxy gets 481 from the sbc and forwards that response back to PSTN gw, which somehow breaks it. AFAICS it's not specified in RFC what should the behavior look like when getting both a 200 and error-class response for the INVITE (quotes are most welcome!). IMO it would be more correct to absorb BYE in proxy but I see a big problem here: branch_2 can even ring for 5 minutes and it's not feasible for proxy to have a wt-timer that long. Also it's not possible to inform the gw that early dialog has cleared as soon as we receive 200/487 from branch_1. So I'm not sure which party is at fault and if we can workaround that somehow in the Kamailio. Any thoughts?

1 1

← Newer
1
2
3
4
5
6
7
8
...
14
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users February 2012