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37517 discussions

wt_timer, delayed replies and sendto error
by Andreas Granig 20 Sep '11

20 Sep '11

Hi, Another funny thing I encountered lately is this: Client sends INVITE via kamailio (via dispatcher) to sems, gets back 100/200, sends another couple of INVITE with same CSeq (looks like retransmissions), gets back more 200, then sends ACKs for each of the 200 it got. Looks like a quite broken client to me, but anyways. The interesting thing is that after >5000ms (which is bigger then tm.wt_timer), another 200 is sent from sems to kamailio to the client. I see the 200 coming to kamailio in a network trace, but not an according log message in the kamailio logs. What I see though is this: Sep 14 13:51:26 sp2 /usr/sbin/kamailio[15950]: ERROR: <core> [udp_server.c:586]: ERROR: udp_send: sendto(sock,0x97ec10,1055,0,1.2.3.4:58620,16): Invalid argument(22) Sep 14 13:51:26 sp2 /usr/sbin/kamailio[15950]: ERROR: <core> [forward.h:149]: msg_send: ERROR: udp_send failed The 1.2.3.4:58620 is obviously taken from the Vias, which looks like this: Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;branch=xyz Via: SIP/2.0/UDP 10.1.1.9:58620;received=1.2.3.4;rport=58620;branch=abc The top-most Via is itself, the second is the one which reflects the destination address in the error message above. So I'm wondering whether there is some issue with finding the proper socket after wt_timer is expired (note that the leg client<->kamailio uses a different socket (with public ip) than kamailio<->sems (which uses 127.0.0.1)). Or any other idea where this error could come from? It's actually an educated guess that this could be related to wt_timer, but I don't know what else it could be. Andreas

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Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 20 Sep '11

20 Sep '11

Hey Phillip, you probably forgot to keep the mailing list in CC, I'll do that for you. On 19.09.2011 08:43, Phillman25 Kyriacou wrote: > You are right. > > I added the function dlg_manage(); when there is a BYE or CANCEL and now > its working perfectly. Before i had this function defined only on > INVITES. The strange thing is that it was working normally before when > dlg_manage(); was defined only for INVITE messages in the route block. So you are saying that before, you called dlg_manage() on INVITEs only but now, you need that function on both INVITE and BYE/CANCEL messages? That'd still qualify as a bug as the trigger to tracking the dialog (i.e., dlg_manage()) is required only once. After that, the dialog module makes sure that all dialog-related messages are properly processed. The only thing that's necessary for sequential requests to be tracked correctly is a call to loose_route() on such messages. If you are sure that your Kamailio configuration didn't change between 3.1.2 and 3.1.5 w.r.t. dialog tracking or anything that could affect it, I'd be willing to take a closer look at this issue (giving you can provide me with more information on your scenario, like call flow examples or similar). Cheers, --Timo > The calls that were suppose to be terminated before where in state 3 to > answer your question. However, i still see calls in state 3 now but they > are valid. > > Regards > Phillip > > On Fri, Sep 16, 2011 at 8:01 PM, Timo Reimann <timo.reimann(a)1und1.de > <mailto:timo.reimann@1und1.de>> wrote: > > Hey Phillip, > > after looking closer at what has been reported to me initially, your > case may be different after all. > > Could you possibly use "kamctl fifo dlg_list" to check what state the > majority of dialogs that should be terminated in your opinion are in? > Being given the reference counter should be useful in debugging this as > well. > > > Thanks, > > --Timo > > > > On 16.09.2011 14:22, Phillman25 Kyriacou wrote: > > On Fri, Sep 16, 2011 at 3:09 PM, Timo Reimann > <timo.reimann(a)1und1.de <mailto:timo.reimann@1und1.de> > > <mailto:timo.reimann@1und1.de <mailto:timo.reimann@1und1.de>>> wrote: > > > > Hey Phillip, > > > > > > On 16.09.2011 13:35, Phillman25 Kyriacou wrote: > > > Hello > > > > > > I have been facing an issue where the dialog module is showing > > calls as > > > being active when in fact the call has already been > completed long ago > > > and this is giving wrong number of concurrent calls on our > SNMP work > > > station (CACTI) when polling the data from Kamailio. I > realized this > > > only started occurring after i upgraded from 3.1.2 to 3.1.5, has > > anyone > > > experienced the same issue? > > > > I was *just* being notified of issues concerning dialogs not being > > deleted. Working on this right now to report back soon. > > > > Thanks for the note! > > > > > > Cheers, > > > > --Timo

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Add client certificate checking against a CRL in opensips
by tomsc 20 Sep '11

20 Sep '11

Hi everyone, I'd like to check that a client certificat is revoked or not against a crl. Actually, opensips use context SSL_CTX. How can I do with this context? I do this change to load the crl : load_crl(SSL_CTX * ctx, char *filename) { LM_DBG("entered load crl\n"); X509_STORE *pStore = SSL_CTX_get_cert_store(ctx); X509_LOOKUP* plookup = X509_STORE_add_lookup(pStore, X509_LOOKUP_file()); if (!X509_load_crl_file(plookup, filename, X509_FILETYPE_PEM)){ LM_ERR("unable to load certificate file '%s'\n", filename); return -1; } LM_DBG("'%s' successfuly loaded\n", filename); return 0; } But I dont know how to add flags, is it the right solution? Thanks all -- View this message in context: http://old.nabble.com/Add-client-certificate-checking-against-a-CRL-in-open… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

Lazy evaluation in if expressions?
by Morten Isaksen 20 Sep '11

20 Sep '11

Does Kamailio use lazy evaluation in if expressions? Like if ($var(a) == 2 && $var(b) ==3) { ... } then $var(b) == 3 is only evaluated if $var(a) == 2 is true -- Morten Isaksen

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directed media problem when video is included in sdp
by Juha Heinanen 20 Sep '11

20 Sep '11

when sems receives voice call invite from ua behind nat that has direction:active in its sdp, like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1. 68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Attribute (a): direction:active sems includes direction:passive in 200 OK and starts to correctly send rtp packets to the ip address/port of received rtp packets. however, then sems receives a voice+video call invite like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1.68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Description, name and address (m): video 8600 RTP/AVP 34 Bandwidth Information (b): AS:300 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:34 H263/90000 Media Attribute (a): direction:active it does not include direction:passive in 200 OK and keeps on sending rtp packets to 192.168.1.68 even after receiving first voice rtp packet from public ip of the ua. is this a bug in sems or is it a bug in sr nathelper_k fix_nated_sdp("1") implementation, i.e., does that function place direction:active line to only one place, when it should place it in two places (in both voice and video media descriptions)? -- juha

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Module db_text
by Bruno Bresciani 20 Sep '11

20 Sep '11

Hi All, I'm starting kamailio version 3.1.2 with db_text module but when it tries to load route table, the following error is generated: * "ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe" * Someone Can Help me to understand why this error is happening? Below is part of kamaio's trace. Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:208]: _tbc->name: route Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:209]: _tbc->nrcols: 9 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:210]: _nc: 8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:238]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_res.c:69]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0x82e23f0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x82e3798 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:165]: allocate 32 bytes for result types at 0x82e37c0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[0] at 0x82e2438 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[1] at 0x82e2390 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[2] at 0x82e23d0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[3] at 0x82e23e0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[4] at 0x82e37e8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[5] at 0x82e37f8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[6] at 0x82e3808 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[7] at 0x82e3818 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:184]: allocate 8 bytes for rows at 0x82e3828 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_row.c:119]: allocate 160 bytes for row values at 0x82e3838 Sep 19 16:13:01 sswpst00 kamailio: ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe Best Regards

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Re: [SR-Users] rtpproxy VS Turn
by Mikael Williams 19 Sep '11

19 Sep '11

> > ICE is not supported as I select the Media Proxy (which is TURN) manually > in some Nat cases...Even note that I do not have peer to peer calls between > 2 endpoints, just have VOIP to PSTN calls...All my VOIP customers supports > STUN/TURN but without ICE > > One more thing: I would like to ask how rtp proxy help to bypass NAT else > than working as relay server...Does it have any other job like find the NAT > type or check blocked port or something? > > Regards > > > On Fri, Sep 16, 2011 at 5:50 PM, Klaus Darilion < > klaus.mailinglists(a)pernau.at> wrote: > >> STUN alone is not 100% sufficient, only ICE+TURN is reliable. Thus, only >> if all your clients support ICE+TURN then you can drop rtpproxy. >> >> klaus >> >> >> On 16.09.2011 16:47, Mikael Williams wrote: >> >>> Dear All, >>> >>> I would like to ask please if and in case I'm using STUN/TURN for Nat >>> traversal to drp the rtpproxy from my solution as NAT traversal and >>> relay server as I'm using SYUN for NAT traversal and TURN as relay in >>> case of Symmetric NAT >>> >>> Regards >>> >>> >>> ______________________________**_________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >

2 1

Re: [SR-Users] sipp to kamailio doesn't match ACK [SOLVED]
by Uri Shacked 19 Sep '11

19 Sep '11

Hi, I got the same problem. Do you have the XML SIPp file for example? 10x Uri

1 0

Dialog module showing calls that have already been terminated
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Hello I have been facing an issue where the dialog module is showing calls as being active when in fact the call has already been completed long ago and this is giving wrong number of concurrent calls on our SNMP work station (CACTI) when polling the data from Kamailio. I realized this only started occurring after i upgraded from 3.1.2 to 3.1.5, has anyone experienced the same issue? my config: ======== loadmodule "dialog.so" . . . . # ----------------- 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", "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) . . . . route { # MANAGE ALL DIALOGS #=================================================== if (is_method("INVITE")) { if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = 12; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); } Thanks Phillip

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kamailio debian/ubuntu support
by Jon Bonilla 16 Sep '11

16 Sep '11

Hi all with the release of kamailio 3.1.5, I've done a small change to the debian repositories: Nightly builds of kamailio_3.0 branch have been dropped and the repository has been erased. There is no development in this branch and it has no sense to keep nightly builds for it. Those (few) people using kamailio_3.0-nightly repositories are advised to move their sources.list to the stable 3.0 repository of upgrade to 3.1 or master branches. kamailio 3.0 will continue to be packaged if any new stable version is released for etch, lenny, squeeze and lucid. For 3.1 branch, nightly builds and stable repos will be provided for lenny, lucid and squeeze When the time for 3.2 (master branch at the moment) comes, I'll add Wheezy support. Not sure about Ubuntu distro support. Should we add support to Ubuntu 11.10 or wait until the lts 12.04 is released? Any opinions welcomed cheers, Jon

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