- sr-users - kamailio.org

Need to confirm the changes for the Kamailio.cfg
by varun pratapsingh 27 Mar '14

27 Mar '14

HI All, This is my first post to this community and I am new user of the Kmailio. I want to use the Kamailio and RTPProxy on the same machine. I have installed and rtpproxy succesfully and I have started both the kamailio and RtpProxy. My RtpProxy is listening on localhost:7722 port of my local machine. Now this machine is on the public IP and I want that all sip and RTP passes though this machine from SIP Softphones before going to the SIp Server. This means the machine on which I am running the Kamailio and RTPProxy will act like a SIP proxy from my SIP Softphones through which the SIP and RTP relays. SIP is relaying fine with defult kamailio.cfg but can anyone let me know the changes in Kamailio.cfg so that my RTP also passes or rleays through this Kamailio + RTPProxy machine. Thanks in advance. Thanks and Regards Varun

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Preventing information about my sip network
by Arya Farzan 27 Mar '14

27 Mar '14

I'm concerned about others reverse engineering their way into my project's sip network. Is there anyway to prevent others from finding out that the SIP protocol is being used and prevent others to reverse engineer their way into my sip network?

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Rewrite SIP INVITE
by Mickael MONSIEUR 26 Mar '14

26 Mar '14

Hello, I installed kamailio and asterisk with the tutorial of asipto. For alias numbers I configured the module alias_db. Everything works because Asterisk outgoing call is directed at Kamailio, then Kamailio is sending to the alias registered (table location) Only the number alias appears in the "To" (02XXXXXX) and not in the INVITE URI. It only shows "s" ... is it possible to force writing 02XXXXXX instead of "s" ? Example: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212@sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX@kamailioIP:5060> Contact: <sip:1053212@asteriskIP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71(a)sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Because without it, Asterisk servers behind Kamailio not will route the call to the correct extension but to the "s". Asterisk ignores the "To" apparently this is strange ... Thank you, Mickael

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mediaproxy-ng, invalid header version
by Mihai Marin 26 Mar '14

26 Mar '14

Hellor Sirs, Sir Richard, I saw some updates in the last 2 weeks that were working with Firefox - I also did some tests as it was working. Now, I tried to get the latest version and I'm getting the following error: mediaproxy-ng[2747]: Got valid command from 127.0.0.1:35127: answer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-29.0a2 13431 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:8a9a3649#015#012a=ice-pwd:80b3b796ecdd0addb079fc531c7e0afa#015#012a=fingerprint:sha-256 A3:D7:1F:F6:60:1D:91:27:64:89:2E:09:CE:42:19:DB:7E:5F:02:06:A3:DA:E0:26:0F:F7:30:4C:1E:65:86:2B#015#012m=audio 50990 RTP/SAVPF 109 101#015#012c=IN IP4 188.215.94.132#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=recvonly#015#012a=setup:active#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50990 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50990 typ srflx raddr 192.168.0.5 rport 50990#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50991 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50991 typ srflx raddr 192.168.0.5 rport 50991#015#012a=rtcp-mux#015#012m=video 50992 RTP/SAVPF 120#015#012c=IN IP4 188.215.94.132#015#012a=rtpmap:120 VP8/90000#015#012a=sendrecv#015#012a=rtcp-fb:120 nack#015#012a=rtcp-fb:120 nack pli#015#012a=rtcp-fb:120 ccm fir#015#012a=setup:active#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50992 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50992 typ srflx raddr 192.168.0.5 rport 50992#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50993 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50993 typ srflx raddr 192.168.0.5 rport 50993#015#012a=rtcp-mux#015#012", "flags": [ "force", "trust-address" ], "replace": [ "origin", "session-connection" ], "call-id": "ufdo7gas8lt6jpco60b1", "received-from": [ "IP4", "188.215.94.132" ], "from-tag": "nj7tuhg5hj", "to-tag": "7lpqrsa1eb", "command": "answer" } mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1] Returning to SIP proxy: d3:sdp1463:v=0#015#012o=Mozilla-SIPUA-29.0a2 13431 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:8a9a3649#015#012a=ice-pwd:80b3b796ecdd0addb079fc531c7e0afa#015#012m=audio 30002 RTP/SAVPF 109 101#015#012c=IN IP4 93.187.138.212#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50990 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50990 typ srflx raddr 192.168.0.5 rport 50990#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50991 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50991 typ srflx raddr 192.168.0.5 rport 50991#015#012a=recvonly#015#012a=rtcp:30002#015#012a=rtcp-mux#015#012a=setup:active#015#012a=fingerprint:sha-1 CA:7D:5B:F1:C8:1A:05:8E:9B:88:0F:36:9B:C4:7A:60:A3:B5:02:A4#015#012a=candidate:kgDINMdmuvprUMd6 1 UDP 2130706432 93.187.138.212 30002 typ host#015#012m=video 30006 RTP/SAVPF 120#015#012c=IN IP4 93.187.138.212#015#012a=rtpmap:120 VP8/90000#015#012a=rtcp-fb:120 nack#015#012a=rtcp-fb:120 nack pli#015#012a=rtcp-fb:120 ccm fir#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50992 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50992 typ srflx raddr 192.168.0.5 rport 50992#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50993 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50993 typ srflx raddr 192.168.0.5 rport 50993#015#012a=sendrecv#015#012a=rtcp:30006#015#012a=rtcp-mux#015#012a=setup:active#015#012a=fingerprint:sha-1 CA:7D:5B:F1:C8:1A:05:8E:9B:88:0F:36:9B:C4:7A:60:A3:B5:02:A4#015#012a=candidate:kgDINMdmuvprUMd6 1 UDP 2130706432 93.187.138.212 30006 typ host#015#0126:result2:oke mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30000] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30004] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30002] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30006] Error parsing RTP header: invalid header version I was dreaming one week ago when I saw Firefox working with chrome :)? Do you know something about this error? Thank you. Best regards, Mihai M

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Using Kamailio with a SIP Trunk
by Rizwan Khan 26 Mar '14

26 Mar '14

I want the following setup: 1 Kamailio server to handle internal calls (A/V), IM and Presence. 1 Asterisk or any other way to communicate with an NGN where I will create the SIP Trunk to route calls outside of the network. Is this the right approach or there is a way to directly communicate with the NGN to make a SIP trunk by using some external modules. Any guidelines will be highly appreciated. Rizwan Khan

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Memory leak when using db_unixodbc, help tracking it down (try 2)
by Alex Villacís Lasso 26 Mar '14

26 Mar '14

I have a CentOS 6 installation with the following packages installed from the RPM build service from Kamailio: kamailio-unixodbc-4.1.2-1.1.x86_64 kamailio-4.1.2-1.1.x86_64 kamailio-presence-4.1.2-1.1.x86_64 kamailio-utils-4.1.2-1.1.x86_64 I am also using db_unixodbc for all database accesses (with MySQL backend). The problem is that after a few hours, a single Kamailio process (never more than one) starts complaining that it has run out of memory: 9(11479) ERROR: uac [uac_reg.c:638]: uac_reg_tm_callback(): got sip response 408 while registering [admin_pbx.elastix.com] 9(11479) ERROR: db_unixodbc [dbase.c:335]: db_unixodbc_fetch_result(): no memory left 9(11479) ERROR: <core> [db_query.c:502]: db_fetch_next(): unable to fetch next rows 9(11479) ERROR: db_unixodbc [dbase.c:224]: db_unixodbc_free_result(): invalid parameter value 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result I have attached the stderr output when I kill the offending process. From what I can see, the memory leak somehow involves database allocations that fail to be freed, consuming more than 3 MB out of 4 MB allocated: 9(11479) NOTICE: <core> [main.c:857]: sig_usr(): Memory still-in-use summary (pkg): 9(11479) NOTICE: qm_sums: summarizing all alloc'ed. fragments: 9(11479) NOTICE: qm_sums: count= 1 size= 10240 bytes from mi_fifo: mi_writer.c: mi_writer_init(57) 9(11479) NOTICE: qm_sums: count= 378 size= 3029384 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(333) 9(11479) NOTICE: qm_sums: count= 1 size= 4096 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(325) 9(11479) NOTICE: qm_sums: count= 1 size= 16 bytes from <core>: sr_module.c: init_modules(1002) 9(11479) NOTICE: qm_sums: count= 1 size= 56 bytes from textops: textops.c: hname_fixup(1634) 9(11479) NOTICE: qm_sums: count= 1 size= 32 bytes from rr: rr_cb.c: register_rrcb(63) I have the setup to compile my own RPMS, and I intend to track down the memory leak on my own. However, I would like your help in focusing my search, on whether this is a known issue, or where in db_unixodbc should I look.

2 3

Regarding SRV lookup
by Swetha Raj 26 Mar '14

26 Mar '14

Hi All, I am trying to communicate between two sip servers.The communication from external sip server user to kamailio user works but message from kamailio to external sip user is not reaching. I get the following error in Jitsi "The above message could not be delivered A network problem occurred. Please check your network configuration and try again. Error was: 408 Request Timeout" Does kamailio do SRV lookup ? How do i force kamailio to use DNS SRV lookup. Any clues ??

1 0

group regex db reload?
by Pete Ashdown 26 Mar '14

26 Mar '14

I'm using the group module to assign groups to users. As the documentation states, "the table content is loaded into memory at startup and all regular expressions are compiled". Having to restart Kamailio is cumbersome every time the table is updated. Is there a method to reload the table via kamcmd? Also, I had no luck in getting the non-regex method of group to recognize a table entry, for example, 159(a)mydomain.com was split into 159 and mydomain.com, yet it get_user_group wouldn't match it. It was only when I started using the regex table that I had success.

2 1

Qualify from Asterisk to Kamailio's peers
by Alexandr Usov 25 Mar '14

25 Mar '14

Peers (from Internet behind NAT) registered on Kamailio (local ip 192.168.182.1), calls from/to routed via Asterisk (192.168.182.24). Can't use qualify info: <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog ' 1efe8f023a80bfb343495d5c4f20ea35@192.168.182.24:5060' Method: OPTIONS <--- SIP read from UDP:192.168.182.1:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.182.24:5060;branch=z9hG4bK4a817c26;rport=5060 From: "asterisk" <sip:asterisk@192.168.182.24>;tag=as34bfcba7 To: <sip:192.168.182.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2d0e Call-ID: 753402f55d9b5ceb6dc402b93e206025@192.168.182.24:5060 CSeq: 102 OPTIONS Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 <------------->

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Unable to make calls after integration with Freeswitch
by mark li 25 Mar '14

25 Mar '14

Hi there. I had a working Kamailio proxy working where i could call from one polycom test phone to another. But then I tried to integrate freeswitch with it... (following the article found at: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc as a baseline. ) Now when I dial a phone, it's just silent for a few seconds... after which i get a busy tone. I did a tcpdump and have been reviewing the file in wireshark. It looks like my two test phones are registered properly... but as you can see, the INVITE is not getting an OK back. It just keeps on retrying the invite. I'm not sure how to go about troubleshooting this. Any suggestions on where to start would be appreciated. In the meantime, I'm googling around to see if I can find anything that might help me figure this out. thanks.

1 0

dialog is not deleted when BYE received
by Tuan Viet Nguyen 25 Mar '14

25 Mar '14

Hello, I disable the rr module and only use SIP elements for dialog match modparam("dialog", "dlg_match_mode", 2) Upon receipt of a BYE request, the request is correctly forwarded and the call ended but the dialog still exists with ref_count = 2 kamctl fifo dlg_list dialog:: hash=731:5167 state:: 3 ref_count:: 2 timestart:: 1394625569 timeout:: 29082806 callid:: 77dd7249530067080de983411e94569c@10.100.8.12:5060 from_uri:: sip:10.100.8.12:5060 from_tag:: as03e52e3f caller_contact:: sip:10.100.8.12:5060 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.100.8.94:5060 callee_bind_addr:: udp:10.100.8.94:5060 Do you have any idea ? Should I remove manually the dialog using a module function? Thanks for your help Best regards,

1 1

Siremis v4.1.0 Released
by Elena-Ramona Modroiu 25 Mar '14

25 Mar '14

Hi, a new release of Siremis is out - v4.1.0 - web management interface that is compatible out of the box with Kamailio v4.1.x series. More details, including link to install tutorial, are available at: - http://siremis.asipto.com/2014/03/25/siremis-v4-1-0-released/ Regards, Ramona

1 0

crash dialog module
by pars3c 25 Mar '14

25 Mar '14

Hi, i have installed the new version ok kamailio 4.1.2. I don't know why Kamailio crash writing this ob the log: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7fc3ab755eb0), called from dialog: dlg_hash.c: dlg_update_cseq(591), first free dialog: dlg_hash.c: dlg_update_cseq(591) - aborting Anybody can help me to undertand what is the problem? thanks

3 2

Ldap auth
by Slava Bendersky 24 Mar '14

24 Mar '14

Hello Everyone, I am trying compare domain part of uri with ldap query result, getting some syntax warning 1. arn_at(): warning in config file /etc/kamailio/kamailio-ldap.cfg, line 992, column 17-39: constant value in if(...) 2. 3. 4. ldap_result("sipExtension/$avp(extension)"); 5. ldap_result("sipDomain/$avp(domain)"); 6. ldap_result("password/$avp(password)"); 7. 8. } 9. 10. if ("$avp(s:domain)" =~ "$fd") { 11. xlog("L_INFO", "Not alllowed $fd"); 12. sl_send_reply("403","Not allowed $fd"); 13. exit; 14. } 15. any help thank you

2 2

presence bug
by Kelvin Chua 24 Mar '14

24 Mar '14

I would like to report a bug on the presence module kamailio 4.0.5 setting custom table names via modparam("presence", "presentity_table", "kam_presentity") modparam("presence", "active_watchers_table", "kam_active_watchers") modparam("presence", "watchers_table", "kam_watchers") does work, i can see entries on the table when logged in. however these things show up from time to time, seems like it is not looking for the custom table name. /usr/local/sbin/kamailio[20406]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Table 'kam.presentity' doesn't exist /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:429]: db_fetch_query_internal(): unable to query db for fetch /usr/local/sbin/kamailio[20406]: ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages /usr/local/sbin/kamailio[20406]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Table 'kam.presentity' doesn't exist /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query /usr/local/sbin/kamailio[20406]: ERROR: presence [publish.c:227]: msg_presentity_clean(): failed to delete expired records from DB /usr/local/sbin/kamailio[20406]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Table 'kam.watchers' doesn't exist /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query /usr/local/sbin/kamailio[20406]: ERROR: presence [subscribe.c:745]: msg_watchers_clean(): cleaning pending subscriptions /usr/local/sbin/kamailio[20406]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Table 'kam.active_watchers' doesn't exist /usr/local/sbin/kamailio[20406]: ERROR: <core> [db_query.c:291]: db_do_delete(): error while submitting query /usr/local/sbin/kamailio[20406]: ERROR: presence [subscribe.c:2220]: update_db_subs_timer(): deleting expired information from database Kelvin Chua

1 0

Memory issue
by Igor Potjevlesch 24 Mar '14

24 Mar '14

Hello, After couple of weeks without issue, my Kamailio 4.0 has begun to reject calls this week-end. The first error : ERROR: <core> [msg_translator.c:2012]: build_res_buf_from_sip_res(): ERROR: build_res_buf_from_sip_res: out of mem. OR ERROR: <core> [msg_translator.c:2012]: build_res_buf_from_sip_res(): ERROR: build_res_buf_from_sip_res: out of mem Followed by ERROR: <core> [forward.c:812]: do_forward_reply(): ERROR: forward_reply: building failed And sometimes, at the end of the line there is a "needs" in addition: ERROR: <core> [msg_translator.c:2164]: build_res_buf_from_sip_req(): ERROR: build_res_buf_from_sip_req: out of memory ; needs 469. The starting options are: OPTIONS="-P /var/run/$prog.pid -m 256". Regards, Igor.

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Re: [SR-Users] Registration error setting pvar
by Nick Raper 24 Mar '14

24 Mar '14

It seems this happens when we try to set the auth_count back to 0 after a successful registration is the following line: $sht(a=>$au::auth_count) = 0; Nick Raper Voice Engineer T +27 (0) 21 710 8400 C +27 (0) 82 777 0855 hymax.co.za <http://www.hymax.co.za/> <http://www.hymax.co.za/> cid:image001.jpg@01CF3399.D9DDFA50 cid:image006.png@01CF339B.A5608F70 This e-mail and any attached files are intended for the named addressee only. It contains information which may be confidential and legally privileged and also protected by copyright. Unless you are the named addressee (or authorized to receive for the addressee) you may not copy or use it, or disclose it to anyone else. If you received it in error please notify the sender immediately and then delete it from your system. We make every effort to keep our network free from viruses. However you do need to check this e-mail and any attachments to it for viruses, as we can take no responsibility for any computer virus which may be transferred by way of this e-mail. From: Nick Raper [mailto:nick.raper@hymax.co.za] Sent: 24 March 2014 11:57 AM To: 'sr-users-bounces(a)lists.sip-router.org' Subject: Registration error setting pvar Hi, Can anyone assist with troubleshooting this error? It happens on client REGISTER requestes. The registration is successful but we get the following errors: 10(2994) ERROR: <core> [lvalue.c:363]: lval_pvar_assign(): setting pvar failed 10(2994) ERROR: <core> [lvalue.c:416]: lval_assign(): assignment failed at pos: (1075,31-1075,31) version: kamailio 4.1.2 (i386/linux) 6fa599 I am using a modified script and DB from version 3.2.1 as we are looking at upgrading our current live servers to ver 4.1.2. I have attached a full debug log of the register. Regards, Nick Raper Voice Engineer T +27 (0) 21 710 8400 C +27 (0) 82 777 0855 hymax.co.za <http://www.hymax.co.za/> <http://www.hymax.co.za/> cid:image001.jpg@01CF3399.D9DDFA50 cid:image006.png@01CF339B.A5608F70 This e-mail and any attached files are intended for the named addressee only. It contains information which may be confidential and legally privileged and also protected by copyright. Unless you are the named addressee (or authorized to receive for the addressee) you may not copy or use it, or disclose it to anyone else. If you received it in error please notify the sender immediately and then delete it from your system. We make every effort to keep our network free from viruses. However you do need to check this e-mail and any attachments to it for viruses, as we can take no responsibility for any computer virus which may be transferred by way of this e-mail.

1 0

kamailio.org website down?
by Isaac McDonald 24 Mar '14

24 Mar '14

Hello, It looks like kamailio.org is having issues. When I try and access it I get a perpetual "waiting for reply". -- Isaac McDonald SusCall, LLC E-Mail: imcdona(a)suscall.com

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Caller ID name
by malik sherif 22 Mar '14

22 Mar '14

Hello, I set SIP users using kamctl add command with username ( I put the phone numbers) doamin (doamin name) and password. Is their a way to add caller id name? when I make a call , i see the phone number but for caller id name I displayline1. Thank you for your help. Abdul

3 4

ldap auth
by Slava Bendersky 22 Mar '14

22 Mar '14

Hello Everyone, Trying setup ldap authentication. Always getting 401. I tested and all avp getting value also domain part check working, look like challenge www in question. Any help thank you in advance. #!ifdef WITH_LDAP route[LDAP] { if(is_method("REGISTER")) { if(!(is_present_hf("Authorization") || is_present_hf("Proxy-Authorization"))) { # no credentials header - send back challenge auth_challenge("$fd", "1"); exit; } # ldap search ldap_search("ldap://sipaccounts/ou=People,dc=networklab,dc=loc?sipDomain,sipMobileExtension,userPassword?one?(&(objectClass=phonesipuser)(sipMobileExtension=$fU))"); $var(rc) = $rc; if ($var(rc)<0) { switch ($var(rc)) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: sl_send_reply("403", "Not allowed"); exit; } } ldap_result("sipDomain/$avp(domain)"); ldap_result("sipMobileExtension/$avp(s:username)"); if (!ldap_result("userPassword/$avp(password)")) { sl_send_reply("404", "User Not Found"); exit; } if ($fd != $avp(domain)) { xlog("L_INFO", "Got ldap result $avp(domain). For user $avp(s:username) Not allowed $fd"); sl_send_reply("403","Not allowed $fd"); exit; } if (!pv_auth_check("$fd", "$avp(password)", "0", "1")) { auth_challenge("$fd", "1"); exit; sl_send_reply("200", "ok"); exit; } else { sl_send_reply("403","Not allowed"); exit; } if (!is_method("REGISTER|PUBLISH")) { consume_credentials(); } } return; } #!endif

1 0

sqlops mysql URL
by Kelvin Chua 21 Mar '14

21 Mar '14

anybody have idea on how to dynamically access mysql URLs using avp? reason being is, i have the mysql URLs in an htable. how can i avoid using modparam for it? is it doable? Kelvin Chua

1 0

Re: [SR-Users] Module TOPOH: Does not unmask CALL-ID on request BYE in one direction
by Samuel F. 21 Mar '14

21 Mar '14

Hey, No, that is totally correct. What is strange is that it doesn’t trigger on the BYE coming back where direction is (should be) = 1. It never unmasks it when it goes back upstream. I will run a scenario in our setup again on Monday and send you the output. Have a nice weekend! // Samuel

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Memory leak when using db_unixodbc, help tracking it down
by Alex Villacís Lasso 21 Mar '14

21 Mar '14

I have a CentOS 6 installation with the following packages installed from the RPM build service from Kamailio: kamailio-unixodbc-4.1.2-1.1.x86_64 kamailio-4.1.2-1.1.x86_64 kamailio-presence-4.1.2-1.1.x86_64 kamailio-utils-4.1.2-1.1.x86_64 I am also using db_unixodbc for all database accesses (with MySQL backend). The problem is that after a few hours, a single Kamailio process (never more than one) starts complaining that it has run out of memory: 9(11479) ERROR: uac [uac_reg.c:638]: uac_reg_tm_callback(): got sip response 408 while registering [admin_pbx.elastix.com] 9(11479) ERROR: db_unixodbc [dbase.c:335]: db_unixodbc_fetch_result(): no memory left 9(11479) ERROR: <core> [db_query.c:502]: db_fetch_next(): unable to fetch next rows 9(11479) ERROR: db_unixodbc [dbase.c:224]: db_unixodbc_free_result(): invalid parameter value 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result I have attached the stderr output when I kill the offending process. From what I can see, the memory leak somehow involves database allocations that fail to be freed, consuming more than 3 MB out of 4 MB allocated: 9(11479) NOTICE: <core> [main.c:857]: sig_usr(): Memory still-in-use summary (pkg): 9(11479) NOTICE: qm_sums: summarizing all alloc'ed. fragments: 9(11479) NOTICE: qm_sums: count= 1 size= 10240 bytes from mi_fifo: mi_writer.c: mi_writer_init(57) 9(11479) NOTICE: qm_sums: count= 378 size= 3029384 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(333) 9(11479) NOTICE: qm_sums: count= 1 size= 4096 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(325) 9(11479) NOTICE: qm_sums: count= 1 size= 16 bytes from <core>: sr_module.c: init_modules(1002) 9(11479) NOTICE: qm_sums: count= 1 size= 56 bytes from textops: textops.c: hname_fixup(1634) 9(11479) NOTICE: qm_sums: count= 1 size= 32 bytes from rr: rr_cb.c: register_rrcb(63) I have the setup to compile my own RPMS, and I intend to track down the memory leak on my own. However, I would like your help in focusing my search, on whether this is a known issue, or where in db_unixodbc should I look.

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[Sr-users]siptrace duplicate_uri
by Tuan Viet Nguyen 21 Mar '14

21 Mar '14

Hello, I set duplicate_uri to store sip messages in an external server #!ifdef WITH_SIPTRACE modparam("siptrace", "db_url", mysql_conn_data) modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on",1) # disable siptrace modparam("siptrace", "trace_flag",FLT_SIPTRACE) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "duplicate_uri", "sip:siptracesrv:6060") #!endif In the route section setflag(FLT_SIPTRACE); but most of the time I get those error messages in log files when kamailio tries to send the message out /usr/local/sbin/kamailio[11946]: ERROR: <core> [forward.c:220]: get_out_socket(): ERROR: get_out_socket: no socket found /usr/local/sbin/kamailio[11946]: ERROR: siptrace [siptrace.c:1487]: trace_send_duplicate(): can't forward to af 2, proto 1 no corresponding listening socket Do you have any ideas? Thanks for your help,

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Contract support engineer in Americas wanted
by Alistair Cunningham 21 Mar '14

21 Mar '14

Integrics would like to hire a contract support engineer to provide level two support to Enswitch customers. This is a contract, work at home role. It entails: * 10 to 20 hours per week (with the possibility of more in future) answering support questions. This is primarily via web ticketing system, though occasional calls with customers may be required. Questions are submitted by customers' engineers, so are generally highly technical. Many involve substantial investigation, debugging, reading of SIP traces, etc. You will not be dealing with end users or the general public. * Being on call for emergency support up to around 50% of the time. This will normally be daytime in North America as we have staff in other countries to cover night, but may occasionally be night time if necessary. You will be compensated for any callouts. * Optionally, some Perl development. If you're not interested in this, or lack the necessary skills, that's no problem. The following are hard requirements. Please do not reply unless you fulfil all of these: * Strong skills in Linux (Redhat and Debian based distributions), Asterisk, and SIP. * Moderate or better skills in Perl, MySQL, and networking. * At least a basic familiarity with Kamailio and/or OpenSIPS. * Experience of providing technical support to IT professionals. * Strong communication and diplomacy skills. * Be physically located in the Americas. Most of the customers you will be supporting are in the USA and Canada, and expect support during business hours. Please do not reply if located outside the Americas. * Be able to speak and write English fluently, and with a professional demeanour, spelling, and grammar. If you're interested in this role and meet the above requirements, please email me off-list. Please include details of your skills and experience, and a rough indication of hourly rates and availability. -- Alistair Cunningham +1 888 468 3111 +44 20 799 39 799 http://integrics.com/

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