- sr-users - kamailio.org

siremis with postgres
by Tomas Zanet 19 Jun '15

19 Jun '15

Hi all, I need to install Kamailio, and Siremis as well, with Postgres backend. My current versions are the following: - Kamailio 4.2.1 - Siremis 4.2.0 - PHP 5.3.10-1ubuntu3.18 - postgres 9.1.3 To do so, I installed libpq-dev and compiled Kamailio with db_postgres module. After that, I create the Kamailio database using kamdbctl create No problems so far....while I’m facing some problem when I tried to install Siremis with Postgres. To do so, I installed php5-pgsql before and then I did the steps described here: http://kb.asipto.com/siremis:install42x:main#dokuwiki__top After clicking on “Setup Now” and “Next” buttons, I’m facing with the following problem: ERROR: SQLSTATE[08006] [7] FATAL: password authentication failed for user "www-data" FATAL: password authentication failed for user "www-data" Though the username I specified into web page is not www-data, but siremis. As far as I know www-data is a Debian user name under which Apache/PHP is running. Can anyone help me please? Thanks in advance Best Regards, Tomas Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 4

kamcmd
by asterisk asterisk 18 Jun '15

18 Jun '15

I am compiling the kamailio 4.3 in raspberry Pi 2. It compiled and worked. However, kamcmd did not work as it looks for the file /var/run/kamailio/kamailio.ctl which is not present. Similar issue found for kamctl but I fixed it by change the /tmp directory to /var/run/kamailio Did I miss something?

2 1

404 Not found error when login Siremis-4.2.0
by Chathura Senanayake 18 Jun '15

18 Jun '15

Hello, I need a help with siremis-4.2.0. I have installed Kamailio-4.3 and have it running. Then I installed Siremis-4.2.0. When I try to login to the Siremis by using login name "admin" and password "admin", it shows a 404 Not found error message as follow. Not Found The requested URL /siremis/system/general_default was not found on this server. ------------------------------ *Apache/2.4.7 (Ubuntu) Server at 172.16.2.9 Port 80* I have run the #a2enmod > rewrite and Restart the apache2. But the problem is still not solved. Please let me know how to resolve this. *Chathura Senanayake* Associate Software Engineer (DA) hSenid Mobile Solutions *Head Office* (Singapore)*P*: +65 65 332 140*M*: *F*: +65 62 910 023 *R&D* (Sri Lanka)*P*: +94 11 268 3751*M*: +94 71 757 9646*F*: +94 11 268 3951 <http://www.hsenidmobile.com/> <http://www.hsenidmobile.com/> <https://www.facebook.com/hSenidMobileSolutions> <https://twitter.com/hSenidMobile> <http://www.linkedin.com/company/hsenid-mobile-solutions-pvt-ltd> <https://plus.google.com/u/0/116242841237814090103/posts> <http://www.youtube.com/user/hSenidmobile>*www.hSenidMobile.com* <http://www.hsenidmobile.com/>

3 7

Kamailio number of matches of specific character in a string
by Ali Taher 18 Jun '15

18 Jun '15

Hello, I just need to count the number of a specific character in a string. For example I want to count the number of "a" in the following string : "bbbaddaa34" I searched among transformations but didn't find such tool. Any help would be appreciated. Thanks Ali

2 1

using uac_replace_from( )
by Al S 18 Jun '15

18 Jun '15

Hi, I am trying to change only the "display name" of the FROM header using uac_replace_from().However, the value is not being changed (see below) request_route { .... xlog ("L_INFO", "fu=$fu"); uac_replace_from("pbx", ""); xlog ("L_INFO", "fu=$fu"); # authentication route(AUTH); ....} Am I missing anything? Any help in this regard would be greatly appreciated. Thanks,Al

3 2

Re: [SR-Users] loose_route to find transport=tcp
by Ryan Kendrick 17 Jun '15

17 Jun '15

After enabling and deciphering debugging it appears there may be a bug. I also reviewed https://tools.ietf.org/html/rfc5658#section-6 I cross-referenced my pcap to ensure I was looking at the reINVITE and see: Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: rr [loose.c:785]: after_loose(): *Topmost route URI: 'sip:xx.xxx.x.xx;lr;r2=on;ftag=a30a720a;did=b75.65a1;nat=yes' is me* Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [xx.xxx.x.xx] == [xx.xxx.x.xx] Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5061 Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [xx.xxx.x.xx] == [xx.xxx.x.xx] Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5070 (advertise 0) matches port 5061 Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [xx.xxx.x.xx] == [xx.xxx.x.xx] Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5090 (advertise 0) matches port 5061 Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [xx.xxx.x.xx] == [xx.xxx.x.xx] Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5061 (advertise 0) matches port 5061 Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: rr [loose.c:181]: find_next_route(): *No next Route HF found* Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: rr [loose.c:847]: after_loose(): no next URI found There is definitely another Route header immediately below the one found above, but find_next_route() doesn't find it . I added my own debugging to loose.c and if ((_m->last_header->type!=HDR_ROUTE_T) || (_m->last_header==*_hdr)) { LM_DBG("No next Route HF found\n"); LM_DBG("_m->last_header->type: %d\n", _m->last_header->type); return 1; } logs find_next_route(): _m->last_header->type: 12 which is HDR_CONTENTLENGTH_T which is indeed the LAST header in the message. We have done very little work in the Kamailio source...just some database escaping in odbc for things to work properly with our database engine...but unless I'm missing something isn't it very wrong to be looking at the last header right here? I may attempt to figure out the message and/or hdr_field data structures and change it. It may also be that the issue doesn't occur when find_next_route is called with a valid _hdr which does seem to search for the "next" one vs going straight to the final header in the entire message. If this is getting overly complicated for this mailing list please let me know... Ryan On Tue, Jun 16, 2015 at 11:40 AM, Ryan Kendrick <kendrick.ryan.c(a)gmail.com> wrote: > > We are using Kamailio 4.2.5 as a registrar and proxy between many > dispersed end-users of a soft phone app and our calling platform / switch. > > Until now we have used udp exclusively but are trying to introduce tcp > between end-users and Kamailio, leaving udp between Kam and our > switch...while maintaining the ability for some end-users to use udp to Kam. > > With some simple address checks I am able to always send to our switch > over udp. If all end-users used tcp I could send everything else tcp, but I > need to maintain udp support. > > The specific problem I am having is on a reINVITE such as this one from > our platform to the a-leg: > > INVITE sip:xxxxxx@xxxxxxxxxxxxx:42679;user=phone SIP/2.0 > Via: SIP/2.0/UDP > xxxxxxxxxxxxx:5060;branch=z9hG4bK218cc8e12ll5035f67INV6a67885312aad > Max-Forwards: 35 > Route: <sip:xxxxxxxxxxx;lr;r2=on;ftag=daba971c;did=b57.4872;nat=yes> > Route: > <sip:xxxxxxxxxxx:5070;transport=tcp;lr;r2=on;ftag=daba971c;did=b57.4872;nat=yes> > Contact: <sip:xxxxxxxxxx@xxxxxxxxxxxxx:5060> > To: "xxxxxx"<sip:xxxxxx@xxxxxxxxxxxxxxxxxxxxxxxxxxx:5070>;tag=daba971c > From: <sip:xxxxxxxxxx@xxxxxxxxxxxxxxxxxxxxxxxxxxx > :5070>;tag=6a678853-co76461-INS002 > Call-ID: MDI4ZmFjNmZhN2Y1NWE2ZTViNTkyZGUwNWE2YzUzYmU > CSeq: 7646101 INVITE > Allow: > INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO,UPDATE > Content-Type: application/sdp > Date: Mon, 15 Jun 2015 20:10:18 GMT > User-Agent: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx > Content-Length: 262 > > As you might notice, we have rr:enable_double_rr set: > > *There are some situations when the server needs to insert two > Record-Route header fields instead of one. For example when using two > disconnected networks or doing cross-protocol forwarding from UDP->TCP. > This parameter enables inserting of 2 Record-Routes. The server will later > remove both of them. * > > and I believe it is necessary to keep this way. Without it Kamailio > doesn't even see the reINVITE...the switch probably tries tcp and that's > not setup between the two. > > The invite above is sent to the a-leg over udp but I would expect and need > it to be tcp in this case. The reINVITE is part of an existing dialog. We > call loose_route() followed by some simple bflag setting and flag checking, > t_on_reply(), ... then t_relay(). > > I do have a functional workaround but would prefer to avoid such manual > handling by utilizing built-in functionality properly. > > # > # relay the message > # > if(route(TEST_TOGW)) { > if (!t_relay_to_udp()) { > sl_reply_error(); > } > } > else { > if ($(hdr(Route)[-1]) =~ "tcp") { > if(!t_relay_to_tcp()) { > sl_reply_error(); > } > } > else if (!t_relay()) { > sl_reply_error(); > } > } > > I'm not 100% sure how reliable or fast this will be, but it does work so > far in my simple tests. > > Is loose_route supposed to see and use the transport=tcp but isn't for > some reason? It seems like the right thing to do to me. If not, is there > anything else I can/should be doing in the tm and/or rr modules to make > Kamailio realize it needs to send this message over TCP? If not in those > two modules is there some recommended way perhaps via registrar or usrloc > etc. to make Kamailio remember/store when a user is connected via TCP and > be able to do a quick lookup before sending to them? Anything else I'm > missing or not thinking of? > > Please let me know if I can further explain and rest assured any > assistance will be much appreciated!!! >

2 2

Performance Tests/Hard Recommendations kamailio 4.2 -200CPS
by Jonathan Hunter 17 Jun '15

17 Jun '15

Hi Guys, Has anyone recently performed tests on kamailio 4.2 running up to 200 CPS? It would be great to know the hardware requirements for this setup. Thanks Jon

5 8

siptrace: how to log sent ACK within dialog
by Seudin Kasumovic 17 Jun '15

17 Jun '15

Hi, TM forward ACK statelessly (see log below), so there is no call of installed callback on TMCB_REQUEST_(IN)SENT in siptrace module, so I can't see way how to log ACK sent packet. Any work around to get sent ACK logged? DEBUG: sl [sl_funcs.c:388]: sl_filter_ACK(): DEBUG : sl_filter_ACK: to late to be a local ACK! DEBUG: <script>: 408:172.16.23.130:5060 <-- 198.51.100.137:52150 ACK sip:8807@127.0.0.1:5080;transport=udp DEBUG: <script>: 425: start trace ACK DEBUG: siptrace [siptrace.c:1865]: pipport2su(): the port string is 52150 DEBUG: siptrace [siptrace.c:1865]: pipport2su(): the port string is 5060 DEBUG: siptrace [siptrace.c:1677]: trace_send_hep_duplicate(): setting up the socket_info DEBUG: <script>: 542: WITHINDLG: loose_route set du='sip:172.16.23.130:5070 ;lr;ftag=1029663405' DEBUG: <script>: 369: RELAY ACK sip:8807@127.0.0.1:5080;transport=udp [4194308] DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=4 , global msg id=3 , T on entrance=0xffffffff DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=34522, isACK=1 DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: *RFC3261 transaction matching failed* DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found DEBUG: tm [t_funcs.c:279]: t_relay_to(): SER: forwarding ACK statelessly DEBUG: <script>: 821:172.16.23.130:5060 --> 172.16.23.130:5070 ACK sip:8807@127.0.0.1:5080;transport=udp Regards, Seudin Kasumovic

2 2

Getting 407 response when calling out (on bulk registration)
by Al S 17 Jun '15

17 Jun '15

Hi, A bulk registration has been added to the "location" table:*************************** 1. row *************************** id: 113 ruid: uloc-55808d5a-33bc-1 username: pbx domain: NULL contact: sip:pbx@10.10.10.157:5060;bnc received: NULL path: NULL expires: 2015-06-17 10:08:42 q: -1.00 callid: 1-15061(a)10.10.10.157 cseq: 2last_modified: 2015-06-17 09:08:42 flags: 0 cflags: 0 user_agent: SIPp/Win32 socket: udp:10.10.10.160:5060 methods: NULL instance: NULL reg_id: 01 row in set (0.00 sec) An outgoing call (SIPOUT) is made from a 10-digit number of the pbx: Request-Line: INVITE sip:9191111111@example.net:5060 SIP/2.0From: sipp <sip:8181111111@home.net>;tag=1To: <sip:9191111111@example.net:5060>Proxy-Authorization: Digest username="pbx",realm="myhome.net",uri="sip:10.10.10.160:5060",nonce="???",response="??",algorithm=MD5 Even though I changed the value of $fU to "pbx", Kamailio does not authenticate the username and response with a 407 "Proxy Auth Required" response: route[AUTH] {... if (is_method("REGISTER") || from_uri==myself) { # authenticate requests $var(fU) = $fU; $fU = "pbx"; if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } ... $fU = $var(fU); #!endif return;} Is this the correct approach to process outgoing call for bulk registration username?Any help in this regard would be greatly appreciated Thanks,Al

1 0

kamailio sending 408 request time out
by Priyaranjan Nayak 17 Jun '15

17 Jun '15

Hi All, I am using kamailio-4.2.5 for testing setup. After starting the kamailio I am getting below message /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -N 20 -m 512 -M 8 /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -E loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(24471) WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.1.118 0(24471) WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.1.118 Listening on udp: 192.168.1.118:4060 tcp: 192.168.1.118:4060 Aliases: 0(24471) INFO: <core> [tcp_main.c:4748]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(24473) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(24473) INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available 0(24473) INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 0(24473) INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(24473) INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 8(24481) INFO: ctl [io_listener.c:225]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Once I sent the request to kamailio server, I am not getting 408 request time out from kamailio and getting below log 12(24485) ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) 12(24485) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 9(24482) ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) 9(24482) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 12(24485) ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) 12(24485) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 10(24483) ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) 10(24483) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 13(24486) ERROR: <core> [tcp_main.c:4162]: handle_tcpconn_ev(): connect 192.168.1.114:61945 failed 13(24486) ERROR: <core> [tcp_main.c:4162]: handle_tcpconn_ev(): connect 192.168.1.114:19129 failed 13(24486) ERROR: <core> [tcp_main.c:4162]: handle_tcpconn_ev(): connect 192.168.1.114:6050 failed Could you please give me any idea how to resolve this issue ? Thanks Priyaranjan

2 1

http_query failed to perform curl (28) errors
by Armen Babikyan 17 Jun '15

17 Jun '15

Hello, I am running across the same issue as described here: http://lists.sip-router.org/pipermail/sr-users/2015-February/086796.html Was there a resolution to this issue? I come across this issue fairly regularly. Unlike the OP, however, I do not have any luck by retrying my query: when I place my http_query within a while loop and keep 1-second delays between calls, all consistently fail with error 28. After long pauses, http_query()s start to work as expected. I think that previous thread mentions DNS as a possible culprit. Is http_query using libcurl's preferred resolver, or is it using Kamailio's built-in DNS cache? I really doubt my local resolver is cause of the issue here. Any insight is appreciated. Thanks! Armen

2 1

Not found error when Siremis login
by Chathura Senanayake 17 Jun '15

17 Jun '15

Hello, I need a help with siremis-4.2.0. When I try to login the the Siremis, it shows a 404 Not found error message as follow. Not Found The requested URL /siremis/system/general_default was not found on this server. ------------------------------ *Chathura Senanayake* Associate Software Engineer (DA) hSenid Mobile Solutions *Head Office* (Singapore)*P*: +65 65 332 140*M*: *F*: +65 62 910 023 *R&D* (Sri Lanka)*P*: +94 11 268 3751*M*: +94 71 757 9646*F*: +94 11 268 3951 <http://www.hsenidmobile.com/> <http://www.hsenidmobile.com/> <https://www.facebook.com/hSenidMobileSolutions> <https://twitter.com/hSenidMobile> <http://www.linkedin.com/company/hsenid-mobile-solutions-pvt-ltd> <https://plus.google.com/u/0/116242841237814090103/posts> <http://www.youtube.com/user/hSenidmobile>*www.hSenidMobile.com* <http://www.hsenidmobile.com/>

1 0

Kamailio Crash
by Marc Soda 17 Jun '15

17 Jun '15

Is it normal for Kamailio to segfault on a duplicate key in the usrloc DB? Jun 15 18:15:23 vproxy2 /usr/sbin/kamailio[13646]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Duplicate entry 'uloc-557a96c1-3577-4e5f1' for key 'ruid_idx' Jun 15 18:15:23 vproxy2 /usr/sbin/kamailio[13646]: ERROR: <core> [db_query.c:337]: db_do_update(): error while submitting query Jun 15 18:15:23 vproxy2 /usr/sbin/kamailio[13646]: ERROR: usrloc [ucontact.c:800]: db_update_ucontact_addr(): updating database failed Jun 15 18:15:23 vproxy2 /usr/sbin/kamailio[13646]: ERROR: usrloc [urecord.c:368]: wb_timer(): updating contact in db failed (aor: sip305_isibmp) I believe that there should never be a duplicate key. I'm not sure how that's happening either. I have 3 Kamailio boxes in a cluster replicating registrations to each other and using a MySQL Galara cluster for the usrloc DB. This is Kamailio 4.0.6. Thanks, Marc

4 4

loose_route to find transport=tcp
by Ryan Kendrick 16 Jun '15

16 Jun '15

We are using Kamailio 4.2.5 as a registrar and proxy between many dispersed end-users of a soft phone app and our calling platform / switch. Until now we have used udp exclusively but are trying to introduce tcp between end-users and Kamailio, leaving udp between Kam and our switch...while maintaining the ability for some end-users to use udp to Kam. With some simple address checks I am able to always send to our switch over udp. If all end-users used tcp I could send everything else tcp, but I need to maintain udp support. The specific problem I am having is on a reINVITE such as this one from our platform to the a-leg: INVITE sip:xxxxxx@xxxxxxxxxxxxx:42679;user=phone SIP/2.0 Via: SIP/2.0/UDP xxxxxxxxxxxxx:5060;branch=z9hG4bK218cc8e12ll5035f67INV6a67885312aad Max-Forwards: 35 Route: <sip:xxxxxxxxxxx;lr;r2=on;ftag=daba971c;did=b57.4872;nat=yes> Route: <sip:xxxxxxxxxxx:5070;transport=tcp;lr;r2=on;ftag=daba971c;did=b57.4872;nat=yes> Contact: <sip:xxxxxxxxxx@xxxxxxxxxxxxx:5060> To: "xxxxxx"<sip:xxxxxx@xxxxxxxxxxxxxxxxxxxxxxxxxxx:5070>;tag=daba971c From: <sip:xxxxxxxxxx@xxxxxxxxxxxxxxxxxxxxxxxxxxx :5070>;tag=6a678853-co76461-INS002 Call-ID: MDI4ZmFjNmZhN2Y1NWE2ZTViNTkyZGUwNWE2YzUzYmU CSeq: 7646101 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO,UPDATE Content-Type: application/sdp Date: Mon, 15 Jun 2015 20:10:18 GMT User-Agent: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Content-Length: 262 As you might notice, we have rr:enable_double_rr set: *There are some situations when the server needs to insert two Record-Route header fields instead of one. For example when using two disconnected networks or doing cross-protocol forwarding from UDP->TCP. This parameter enables inserting of 2 Record-Routes. The server will later remove both of them. * and I believe it is necessary to keep this way. Without it Kamailio doesn't even see the reINVITE...the switch probably tries tcp and that's not setup between the two. The invite above is sent to the a-leg over udp but I would expect and need it to be tcp in this case. The reINVITE is part of an existing dialog. We call loose_route() followed by some simple bflag setting and flag checking, t_on_reply(), ... then t_relay(). I do have a functional workaround but would prefer to avoid such manual handling by utilizing built-in functionality properly. # # relay the message # if(route(TEST_TOGW)) { if (!t_relay_to_udp()) { sl_reply_error(); } } else { if ($(hdr(Route)[-1]) =~ "tcp") { if(!t_relay_to_tcp()) { sl_reply_error(); } } else if (!t_relay()) { sl_reply_error(); } } I'm not 100% sure how reliable or fast this will be, but it does work so far in my simple tests. Is loose_route supposed to see and use the transport=tcp but isn't for some reason? It seems like the right thing to do to me. If not, is there anything else I can/should be doing in the tm and/or rr modules to make Kamailio realize it needs to send this message over TCP? If not in those two modules is there some recommended way perhaps via registrar or usrloc etc. to make Kamailio remember/store when a user is connected via TCP and be able to do a quick lookup before sending to them? Anything else I'm missing or not thinking of? Please let me know if I can further explain and rest assured any assistance will be much appreciated!!!

1 0

Keep-Alive in dialog "freeing a free fragment"
by Dirk Teurlings - SIGNET B.V. 16 Jun '15

16 Jun '15

Hi, (Kamailio 4.2.3) Working to get dialogs closed and cleaned up nicely I'm looking into the ka_interval and ka_timer parameter of the dialog module. The following is now happening. Scenario: 1. Call gets setup 2. Callee's internet connection drops 3. Call gets terminated (nicely) due to ka_timer/ka_interval 4. Log fills with "freeing a free fragment" after about 10 seconds Digging into this I saw that the dialog is being closed and cleaned properly in step 3. But it fails to remove/clean the timers used for the keep-alive. That's what triggers step 4, which in turn tries to clean the (or some of the) dialog space again. Is this expected behaviour when using the keep-alive options, or am I running into somekind of (known) bug? Thanks in advance for any help. -- Cheers, Dirk Teurlings

2 12

CANCEL request and subsequent 487 response to INVTE configuration problem
by Joao Alves 16 Jun '15

16 Jun '15

Hi, I'm creating a configuration is for a SIP <> HTTP gateway (performing protocol conversion) and thus all SIP messages and responses needs to be generated from the kamailio config file. I'm failing to create the case where the calling user abandons the session, and thus, following the reception of a CANCEL a 200 "OK" needs to be sent to this transaction and later the 487 "Request Terminated" to the SIP INVITE original session. On the traces I see: SIP INVITE (Cseq 1 INVITE) --> 100 Trying (Cseq 1 INVITE) <-- 180 Ringing (Cseq 1 INVITE) <-- SIP CANCEL (Cseq 1 CANCEL) --> 200 Ok (CSeq 1 CANCEL) <-- 487 Request Terminated (CSeq 1 CANCEL) Have then tried several options, trying to force the CSeq 1 INVITE but without success. t_reply_callid("$ci", "$cs", "487", " Request Terminated "); t_reply_callid("$ci", "$rm", "487", "Request Terminated"); t_reply_callid("$ci", "1", "487", "Request Terminated"); For instance, for the last I get the following log. 6(60226) exec: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=705 a=28 n=t_reply_callid 6(60226) DEBUG: tm [t_lookup.c:1715]: t_lookup_callid(): created comparable call_id header field: >Call-ID: 76589ZTJmNTAwZDhlOTZiM2I3MjhhYzllNjgyOWVjZGZmMzk < 6(60226) DEBUG: tm [t_lookup.c:1719]: t_lookup_callid(): created comparable cseq header field: >CSeq: 1 INVITE< 6(60226) DEBUG: tm [t_lookup.c:1722]: t_lookup_callid(): just locked hash index 18668, looking for transactions there: 6(60226) DEBUG: tm [t_lookup.c:1749]: t_lookup_callid(): DEBUG: t_lookup_callid: transaction not found. 6(60226) DEBUG: tmx [tmx_mod.c:500]: t_reply_callid(): Lookup failed - no transaction Could you assist here? Thanks, Joao Joao Alves Solution Architect, Unified Communications +351 214094660 (desk) +351 912783702 (mobile) AMDOCS | EMBRACE CHALLENGE EXPERIENCE SUCCESS This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

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Kamailio license usage
by Alexandru Covalschi 15 Jun '15

15 Jun '15

Maybe it may be an offtopic, but I'm not really into legal issues - so I'm sorry if this message is not fully related to this mailing list. Can I use Kamailio to provide VoIP backend for kind of CRM system in case of SaaS? --- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

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Re: [SR-Users] Kamailio number of LCR rules
by Juha Heinanen 15 Jun '15

15 Jun '15

Henry Fernandes writes: > But we didn¹t want to add hundreds of thousands or millions of extra rules > just to deal with this problem. We didn¹t consider that a good > workaround. OK, I got it. Now priorities choose from equally long prefixes instead of all matching prefixes. I'll take a look at the code to check if the latter could be easily added as an option. -- Juha

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Re: [SR-Users] Kamailio number of LCR rules
by Juha Heinanen 15 Jun '15

15 Jun '15

Henry Fernandes writes: > Let’s say you have the following info in your LCR. > > Prefix, Carrier/Route, Rate > 1306, A, 1.5 > 1306343, A, 1.7 > 1306, B, 1.4 > > If you have phone number 13063431111, the LCR module matches it to the > longest prefix (1306343) and routes the call to Carrier A. However, that > is incorrect. It should match it to Carrier B which is the cheapest. > Carrier B has provided a rate of 1.4 for all calls with prefix 1306. why can't you add rule 1306343, B, 1.4 to solve the problem? -- juha

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Kamailio number of LCR rules
by Ali Taher 15 Jun '15

15 Jun '15

Hello, I'm testing LCR module in Kamailio and everything works as expected so far. Before going further , I just want to ask about how much rule in LCR_rule table and target in LCR_rule_target can Kamailio support . We are dealing with hundreds of millions of rules and targets. Kindly advise if Kamailio can support this number of rules. Thanks Ali

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Removing parameters from user part of URI
by Alex Balashov 15 Jun '15

15 Jun '15

Hi, If I have an RURI like this: INVITE sip:4045551212;param=xyz@domain:port SIP/2.0 which transformations can I best string together in order to remove the ';param=xyz' bit and extract just the core user part value (4045551212)? {uri.user} returns '4045551212;param=xyz'. There are numerous transformations to help one access the parameters. What if I want to exclude the parameters? I've already scoured the available transformations, just hoping a second pair of eyes can tell me I've missed something useful. Cheers, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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SIP-over-Websocket Load Balancing
by Murugan Pandian 15 Jun '15

15 Jun '15

HI, how to handle sip-over-websocket load balancing (WebRTC)

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Websocket Load Balancing
by W5RTC 14 Jun '15

14 Jun '15

HI, how to handle SIP-OVER-Websocket loab balancing (WebRTC)

3 2

ASYNC Module
by Aaron Hamstra 13 Jun '15

13 Jun '15

We just updated our development environment from 4.2.2 to 4.2.5 and started noticing the server sending a 500 I'm terribly sorry, server error occurred (1/TM) error back as soon as the async_route block gets called. I have removed everything from our async route other than an xlog statement trying to determine if something in that route block was causing the 500 to occur. It still happens even with only the xlog. If the async_route is not in the config, then the server does not send the 500 response. We rolled back to 4.2.2 and the problem no longer occurs. I have attached a sip trace of the messages. I believe the relevant parts of our config are below, please let me know if you need more from the config however. Any ideas? async_route("PUSHJOIN", "2"); -------------------------- route[PUSHJOIN] { xlog("L_INFO", "Does it get here?\n"); exit; } Thanks in Advance, Aaron

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lookup("location") question
by Al S 12 Jun '15

12 Jun '15

Hi, lookup("location") functions extracts username from Request-URI and relay INVITE without any username, causing UAC sends a CANCEL message. Is the function suppose to replace uri with "contact" field in the found entry in "location" table? INVITE uri arrives at Kamailio:Request-Line: INVITE sip:8181111111@example.net:5060 SIP/2.0 INVITE uri leaves Kamilio:Request-Line: INVITE sip:10.10.10.1:5060 SIP/2.0 lookup("location") function finding match and returns 1: $rc=1 entry in "location" table for this call:*************************** 1. row *************************** id: 104 ruid: uloc-557ad93a-2d6-1 username: 8181111111 domain: NULL contact: sip:8181111111@10.10.10.1:5060 received: NULL path: NULL Any help in this regard would be greatly appreciated Al,Thanks

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