- sr-users - kamailio.org

sending INVTE with Digest values
by Al S 24 Jul '15

24 Jul '15

I am registering my client to kamailo successfully. client --> Register --> kamailioclient <-- 401 with nonce value <-- kamailioclient --> Register with nonce and md5 response values --> kamailioclient <-- 200ok <-- kamailio However, when the client sends an invite with the same digest values, kamailio sends a 407 request for another challenge: client --> Invite with the same nonce and md5 response values --> kamailioclient <-- 407 <-- kamailio I am thinking the same digest values from register could be used for making calls. Thanks,Al

2 1

Dispatcher module number of calls
by Alberto Sagredo 23 Jul '15

23 Jul '15

Hi Hi have read documentation but it seems dispatcher does not keep how many calls has been dispatched or currently are in any of dispatcher destinations I have take a look to code on: http://lists.sip-router.org/pipermail/sr-users/2012-July/073919.html But it seems to use calls limit per user, I would need to establish a limit per dispatcher destination i order to do not send more calls. Any idea where to look for? I have take a look to DMQ module, that maybe i could to use to spread information of how many calls to every destination but do not know how to extract calls from kamailio that have been dispatched to this destination using disptcher module. Best Regards Alberto

3 5

Kazoo like load balancing
by Jurijs Ivolga 23 Jul '15

23 Jul '15

Hi, First of all I would like to mention, that I'm very new to Kamailio. :) My target is to create simple load-balancing similar what have Kazoo, basically 2 servers will share same domain name. I installed 2 Kamailio 4.3 servers and configured this 2 servers with one domain address. I used almost default config with: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_NAT #!define WITH_NATSIPPING Calls are going through if I register both clients to one server. But all my calls are failing, when clients are registered to different servers with error "404 Not Found". Maybe somebody can give me a hint how to make this work. Thank you! With kind regards, Jurijs

3 19

Re: [SR-Users] Kamailio mod_destroy() ERROR
by Ali Taher 23 Jul '15

23 Jul '15

Any help regarding the below would be appreciated. BR, Ali From: Ali Taher [mailto:ataher@vanrise.com] Sent: Wednesday, July 22, 2015 1:39 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Cc: 'Ali Taher~Vanrise Technical Support' <ataher(a)vanrise.com> Subject: Kamailio mod_destroy() ERROR Hello, I'm getting below error when starting Kamailio. ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Any advice ? Regards, Ali

2 1

Port 5061 Issue
by Ankhit Vivekananda 23 Jul '15

23 Jul '15

Hello, I have configured kamailio 4.0.4 long back. Now, I wanted to enable the usage of port 5061 with TLS. By doing so, the client(video camera) can place outgoing calls fine. But while receiving an incoming call, the message does not get sent out to the other client. Hence, the call fails. I can place and receive calls on port 5060 successfully without any issues. Can anyone please tell me why port 5061 is giving me a problem. Also, as a test, I set up kamailio on another machine(version 4.1) and 5061 works fine in all directions. It's only on the other machine that I have problems with. Appreciate any help! Thank you, Ankhit

2 1

htable replication between several kamailio's through dmq
by José Seabra 22 Jul '15

22 Jul '15

Hello there, I'm using the dmq and htable modules to limit the number of concurrent calls in several kamailio's servers. Every call that one kamailio server receives, kamailio increments a htable variable that is replicated to other kamailios through dmq, but if one of these kamailios fails or is restarted it looses the data from htable variable used to count the number of calls and it is not recovered again when kamailio starts again. Is there any way to recover the data from htable variable when kamailio starts, from another kamailio instances through dmq module? Best Regards -- Cumprimentos José Seabra

2 1

RTP Proxy to WAV Script conversion
by Alberto Sagredo 22 Jul '15

22 Jul '15

Using RTPBreak and sox im able to convert rtpproxy rtp files to wav. Later you could convert to mp3 if liked Just to share with list https://github.com/albersag/rtpproxy-utils Any comment is well appreciated Best Regards

2 1

dmq_is_from_node() issue
by Asgaroth 22 Jul '15

22 Jul '15

Hi All, I have an issue with the dmq_is_from_node() function, whereby it does not detect a node is on the bus if it has been removed from its "table" due to previously failed, i presume, ping requests. I have the following snippet in my main routing block: if(is_method("KDMQ")) { if (dmq_is_from_node()) { dmq_handle_message(); exit; } else { xlog("L_INFO", "KDMQ message recieved from node not on our bus!! ($si:$sp) Dropping."); exit; } } I have the following dmq module parameters enabled for dmq: modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:lb.domain.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) DMQ_ADDRESS is set to each of the addresses in the lb.domain.com for each local kamailio instance lb.domain.com resolves to the following addresses: 10.6.0.109 10.6.0.110 10.6.0.111 If I start up the 1st instance of kamailio and do not start the other 2 within the default ping timout then I get the expected log message of dmq removing the nodes from its table, for example: kamailio[26657]: ERROR: dmq [notification_peer.c:588]: notification_resp_callback_f(): deleting server sip:10.6.0.111:5060 because of failed request kamailio[26657]: ERROR: dmq [notification_peer.c:588]: notification_resp_callback_f(): deleting server sip:10.6.0.110:5060 because of failed request However, if I then start one, or both, of the other nodes, then the KDMQ requests are dropped due to my snippet above, I presume, because dmq_is_from_node() checks its "active" table. See messages below as an example of the above snippet in action: kamailio[26656]: INFO: <script>: KDMQ message recieved from node not on our bus!! (10.6.0.110:5060) Dropping. kamailio[26656]: INFO: <script>: KDMQ message recieved from node not on our bus!! (10.6.0.111:5060) Dropping. Is this expected behaviour from the dmq_is_from_node() function? I would have expected it to always check the addresses resolved in the notification address, or am I using this function in a manner not intended. Any suggestions/tips would be greatly appreciated. Thanks

3 5

RFC: about the self signed tls certificates
by Daniel-Constantin Mierla 22 Jul '15

22 Jul '15

Hello, (cross-posting as it impacts users as well) Currently we generate and install the TLS self-signed certificates when tls module is installed, including them in the debian packages (and I guess in rpms). Debian has a policy of reproducible builds, meaning that from same source tree snapshot the same binary packages should result. Even more important considering the impact on security, it would be better that the certificates are generated locally on the installation server, to be distinct. Right now, people relying on default installation config/certificates (and I guess there are many, at least in testing phase), are exposed to eavesdropping, because the private key is available in public packages. My proposal is to move generation of self signed certificates to kamctl. There can be a kamctl.tls file to be deployed by the tls package (same is done by kamctl.mysql, being part of mysql package), which should add a new group of commands, among them something like: kamctl tls generate-certificate The drawback is that before enabling tls and starting kamailio, one has to run the above command. We can document that in tls module readme and in kamailio.cfg in the comments related to WITH_TLS define. Anyone with comments, pros/cons? Other suggestions on how to address the reproducible builds as well as solve the security issue for the default installation? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

5 4

Kamailio mod_destroy() ERROR
by Ali Taher 22 Jul '15

22 Jul '15

Hello, I'm getting below error when starting Kamailio. ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Any advice ? Regards, Ali

1 0

replace From header username for AUTH
by Al S 22 Jul '15

22 Jul '15

Hi, I was wondering how can I replace From header username to perform the following authentication check with the new value: route[AUTH]......if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "1"); exit;}... Thanks,Al

2 1

kamailio 4.3v partition column name is reserved word in mysq ndbcluster engine
by José Seabra 21 Jul '15

21 Jul '15

Hello Daniel, After update my kamailio to version 4.3 and change the table location as required in migration process, kamailio start giving some errors: [root@ccp1 kamailio]# 35(7637) ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Unknown column 'partition' in 'where clause' 35(7637) ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query that's is because the column name "partition" is reserved word ( http://dev.mysql.com/doc/refman/5.6/en/keywords.html) in mysql ndbcluster engine, so what i did to fix this issue was add a new modparam in modules/usrloc/ul_mod.c file: {"partition_col", PARAM_STR, &partition_col}, with this i can configure what name i want for this column in kamailio module configuration and configure a diferent column name in table location(ex: modparam("usrloc", "partition_col", "partitionusrloc") ), Can you please check out if what i did is the correct way to fix it, in order to avoid damage something else. Best regards -- Cumprimentos José Seabra

2 6

Compiling only one module
by Loic Chabert 21 Jul '15

21 Jul '15

Hello, I'm writing a new module for influxdb stats. I've wrote my code and now i need to compile it. How can i compile it ? I don't want to compile all modules, only this one. Thanks in advance, Regards.

2 3

usage of t_flush_flags() in event_route
by Andrew Pogrebennyk 20 Jul '15

20 Jul '15

Hi Daniel and others, I'm having a problem with acc module if I'm using the event_route/ branch-failure: say, the call comes from the app server and goes to the registered user. We arm the the failure route and per-branch failure route for the 302 redirect from the UA. We explicitly reset the accounting flags because we don't want to account the calls from the app server. The transaction is created implicitly by the t_relay(). Now if the UA responds with the 302 response and we are going to process that, we want to create an acc record for the new target from 302 message because this call may incur additional costs. I'm setting the accounting flags and calling t_flush_flags() but that doesn't work (no accounting record). Any idea if I'm doing something wrong or maybe there's a bug when changing the flags and then calling t_flush_flags from the event_route? Here are the module parameters: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "db_url", PAIR_URL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "cdr_log_enable", 0) FTR, we tried 4.1.6 and 4.1.8. And here is the event route (with flags defined like this: flags FLAG_ACC_DB:1, FLAG_ACC_MISSED:2, FLAG_ACC_FAILED:3, ...): event_route[tm:branch-failure:redirect] { route(ROUTE_STOP_RTPPROXY_BRANCH); if($T_rpl($rs) == 301 || $T_rpl($rs) == 302) { # initialise variables when entering failure route route(ROUTE_INITVARS); # these need to be avps because we need it in reply/failure-route $(avp(s:from_faxserver)[*]) = 0; $(avp(s:to_faxserver)[*]) = 0; $(avp(s:cf_from_pstn)[*]) = 0; $(avp(s:from_pstn)[*]) = 0; $(avp(s:proxylu_from_pstn)[*]) = 0; $(avp(s:lcr_flags)[*]) = 0; $(avp(s:em_call)[*]) = 0; $(avp(s:from_pbx)[*]) = 0; $(avp(s:p_to_device)[*]) = 0; $(avp(s:p_to_group)[*]) = 0; $(avp(s:is_primary)[*]) = 0; # now let's process a 30x $(avp(s:acc_state)[*]) = "cfb"; $(avp(s:orig_acc_caller_user)[*]) = $avp(s:acc_caller_user); $(avp(s:orig_acc_caller_domain)[*]) = $avp(s:acc_caller_domain); $(avp(s:acc_caller_user)[*]) = $avp(s:acc_callee_user); $(avp(s:acc_caller_domain)[*]) = $avp(s:acc_callee_domain); $(avp(s:caller_uuid)[*]) = $avp(s:callee_uuid); $(avp(s:callee_uuid)[*]) = $null; # the $var(no_acc) is 0 at this point but the flags may have been reset # if this is a call from PBX user - we do want accounting for the 302 redirect if(isflagset(FLAG_ACC_DB)) { xlog("L_NOTICE", "++++++ ACC flag is set - [% logreq -%]\n"); } else { xlog("L_NOTICE", "------ ACC flag is NOT set - [% logreq -%]\n"); } setflag(FLAG_ACC_FAILED); setflag(FLAG_ACC_DB); t_flush_flags(); # get last URI from destination-set and set it as R-URI $var(contact) = $T_rpl($ct); $var(contact) = $(var(contact){nameaddr.uri}); if($var(contact) == 0 || $var(contact) == $null) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [% logreq -%]\n"); acc_db_request("480", "acc"); $var(announce_handle) = "callee_tmp_unavailable"; $var(announce_set) = $xavp(callee_real_prefs[0]=>sound_set); $(avp(s:announce_code)[*]) = 480; $(avp(s:announce_reason)[*]) = "Temporarily Unavailable"; route(ROUTE_EARLY_REJECT); } $ru = $var(contact); xlog("L_NOTICE", "Redirect from UAC intercepted - [% logreq -%]\n"); $(avp(s:forwarder_cli_userprov)[*]) = $T_rpl($tU); $(avp(s:forwarder_domain_userprov)[*]) = $T_rpl($td); $var(forward) = 1; $var(redirected_forward) = 1; route(ROUTE_LOAD_CALLER_PREF); route(ROUTE_FIND_CALLEE); } } Thanks. Andrew

2 5

File Transfer
by Abdul Hakeem 20 Jul '15

20 Jul '15

Hello, Is there a support for file transfer between registered SIP clients, If not, how can this be achieved ? Regards, Abdul Hakeem

1 0

Kamailio v4.3.1 Released
by Daniel-Constantin Mierla 20 Jul '15

20 Jul '15

Hello, Kamailio SIP Server v4.3.1 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.3.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.1. For more details about version 4.3.1 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-3-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Missing BYE at the far end
by Kliment Toshkov, Netfinity JSC 20 Jul '15

20 Jul '15

Dear colleagues, We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE. Current setup is the following: PEER IP > KAMAILIO 4.2.2 > (Asterisk PBX 1, 2, 3, …, N) We provision only termination services, which is: 1) peer calls us 2) Kamailio chooses destination using dispatcher module 3) forwards the call to it. Attached is captured trace. Clearly BYE is retransmitted a few times, then request for UPDATE is received but it is not possible to answer it because the call is already hung up (as far as I understand). Kamailio routing file is available upon request. Thank you in advance! Климент Тошков Изпълнителен директор 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

2 1

Planing to release v4.3.1
by Daniel-Constantin Mierla 20 Jul '15

20 Jul '15

Hello, I am considering to release the first patch version in 4.3 series by beginning of next week (Monday or Tuesday, July 20 or 21). If you are aware of issues not reported yet, then open an item on the tracker to be taken care: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
by Ben Fitzgerald 20 Jul '15

20 Jul '15

Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box. However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well. This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers): <--- SIP read from TCP:kamailio:41205 ---> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 To: <sip:40081@asteriskIP> From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0 (x86_64/linux)) Contact: <sip:40081@kamailio:5060> Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to kamailio:5060 (no NAT) Sending to kamailio:5060 (no NAT) <--- Transmitting (no NAT) to kamailio:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailio From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: <sip:40081@asteriskIP>;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER ========================= sip.conf for kamailio trunk: [kamailio-inbound] type=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no ======================== Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

3 7

Double Record-Route to subscriber with multiple locations on different SIP transports
by Andrey Utkin 17 Jul '15

17 Jul '15

Hi! We found that we need double Record-Route when we call from one SIP transport to another (let's say, TCP -> TLS), because without |transport=" hint in Record-Route, mobile Linphone app sends messages (INVITE reply, ACK, BYE) via UDP, even if configured for another transport. record_route_advertised_address() is told to add double RR if needed, automatically, so we use it. But the subscriber (the receiver of call) can have multiple registered locations, let's say, one connected via TCP, and another via TLS. So we need to handle each call branch added by lookup(). But adding record_route_advertised_address() ONLY to branch_route attached by t_on_branch() doesn't execute if there's only one location. Having record_route_advertised_address() just in routing script after lookup() seems wrong because it seems unable to edit it for other branches (I don't remember if I have tested this and have proven a failure, or this is just my speculation). So we have added it in both places - right under lookup(), and in branch_route. But in this case we have an error logged: ERROR: rr [rr_mod.c:337]: w_record_route_advertised_address(): Double attempt to record-route Could anyone explain what would be the most correct solution? What is intended behaviour of branch_route? Is it omitted from execution on "primary" branch of call? Is there any parameter which could help? Thanks in advance. -- Andrey Utkin

3 7

Dialogue restore after initial processing
by Joao Alves 17 Jul '15

17 Jul '15

Hi, I'm trying to restore a SIP session dialogue context and found dlg_get(callid, ftag, ttag) function for that. However, I cannot access the To tag. I noticed on the 180 RINGING there was already an self-generated included on this response, but I also could not access its value. When receiving the SIP INVITE request I used the dlg_manage() and the context was initially stored. if(is_known_dlg()) { xlog("Request $rm from $ci is in-dialog\n"); xlog(">>From tag is $ft and To tag is $tt\n"); # Needed to later restore the dialog $shv(ftag)=$ft; $shv(ttag)=$tt; } 15(49100) ERROR: <script>: Request INVITE from 76589ZWQ5NzIyYjMzNDI1YjVjM2E5ZWU5NjhkYzEyNjE3MzQ is in-dialog 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1021 a=25 n=xlog 15(49100) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter 15(49100) ERROR: <script>: >>From tag is 8fd06474 and To tag is <null> Since there was a self-generated one (after the 180 RINGING being sent), I rechecked the $tt and $dlg(to_tag) values and both still returned null values. 15(49100) DEBUG: tm [t_lookup.c:1083]: t_check_msg(): DEBUG: t_check_msg: T already found! 15(49100) DEBUG: <core> [tcp_main.c:2196]: tcpconn_send_put(): send from reader (49100 (15)), reusing fd 15(49100) DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... 15(49100) DEBUG: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7effdf7da860 n=396 fd=5 15(49100) DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf= SIP/2.0 180 ringing Via: SIP/2.0/TCP 192.168.1.5:58751;branch=z9hG4bK-524287-1---a64e0a42199ee419;rport=61134;received=85.138.200.8 To: <sip:1001@decasa.website>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f6a3 From: "1000"<sip:1000@decasa.website>;tag=8fd06474 Call-ID: 76589ZWQ5NzIyYjMzNDI1YjVjM2E5ZWU5NjhkYzEyNjE3MzQ CSeq: 1 INVITE Server: kamailio (4.3.0 (x86_64/linux)) Content-Length: 0 ..., shmem=0x7effdf7e1e00: SIP/2.0 180 ringingght(): DEBUG: reply sent out. buf=0x7effec023bf8: SIP/2.0 180 ringing 15(49100) DEBUG: tm [t_reply.c:657]: _reply_light(): DEBUG: _reply_light: finished 15(49100) DEBUG: sl [sl.c:272]: send_reply(): reply in stateful mode (tm) 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1031 a=25 n=xlog 15(49100) DEBUG: dialog [dlg_hash.c:640]: dlg_lookup(): ref dlg 0x7effdf7e1bf0 with 1 -> 2 15(49100) DEBUG: dialog [dlg_hash.c:642]: dlg_lookup(): dialog id=5769 found on entry 128 15(49100) DEBUG: dialog [dlg_hash.c:858]: dlg_unref(): unref dlg 0x7effdf7e1bf0 with 1 -> 1 15(49100) ERROR: <script>: Dialog To Tag: <null> <== xlog("Dialog To Tag: $dlg(to_tag)\n"); 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1033 a=25 n=xlog 15(49100) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter 15(49100) ERROR: <script>: >>From2 tag is 8fd06474 and To tag is <null> Many thanks, Joao Alves Solution Architect, Unified Communications 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

2 3

Configuring Kamailio as SIP Proxy
by Djamel Bahamid 16 Jul '15

16 Jul '15

Hi, I have terminals of video conference which are in public IP ( *DMZ*). I installed kamailio (4.2) which works correctly, it is also in IP pulique (*DMZ*). I wish configured the /kamailio.cfg/ file, so that kamailio works in *proxy SIP*. And declared my CODEC of visoconférence in URI, example: 'DR London(a)domaine.fr'. In what part of the file ( *kamailio.cfg* ) do I have to make these modifications ? This to avoid the flooding. How possibly write the rule of corresponding ACL. Thank you for your help. Best Regards,

3 4

interconnect two Kamailio server
by Kai Ohnacker 16 Jul '15

16 Jul '15

Hello, I have a setup with 2 Kamailio server and their own database. Is it possible to connect these 2 Kamailio server, that the client from the one can make a call to the other? If yes, how could this be realized? Many Thanks. Cheers, Karl --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com

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Error forwarding to SCSCF
by Jan Devai 16 Jul '15

16 Jul '15

*Hi Experts,* *I would like to ask you if some of you have already come across this problem.* *We are integrating Kamailio all in one Solution (P/I/S-CSCF having all same IP only communicating between each other on different ports) taken from Carsten Bock example against an external HSS.* *The flow is as follows:* UE HSS P-CSCF I-CSCF S-CSCF --------------------------Sip Register--------------------------> ----------Sip Register---------> <---------------------------------UAR-------------------------------- ---------------------------------UAA--------------------------------> --------------SIP Register-------> *The last step "SIP Register" from I-CSCF to S-CSCF is not happenning for some reason.* *The Received AVP in UAA is "First Time Registration 2001 code".* *Now we do not know whether this UAA answer is OK, or some other URI is badly defined which could explain why I-CSCF has trouble contacting S-CSCF.If someone have some logs, traces which could be used as a reference for this scenario would be greatly appreciated.* *Find theses messages we get in the shown logs below:* *Thank you* *JanThis file /rtc/kamailio_icscf/kamailio.cfg is a function which sends message to SCSCF, see below:* route[REG_UAR_REPLY] { #xlog("$$avp(s:uaa_return_code) = $avp(s:uaa_return_code)\n"); #this is async so to know status we have to check the reply avp switch ($avp(s:uaa_return_code)){ case 1: #success if (I_scscf_select("0")){ t_on_failure("register_failure"); t_on_reply("register_reply"); #now relay to appropriate SCSCF if (!t_relay()) { t_reply("500", "Error forwarding to SCSCF"); } } else {#select failed I_scscf_drop(); t_reply("500", "Server error on SCSCF Select (UAR)"); } break; case -1: #failure xlog("L_ERR", "UAR failure - error response sent from module\n"); break; case -2: #error xlog("L_ERR", "UAR error - sending error response now\n"); t_reply("500", "UAR failed"); break; default: xlog("L_ERR", "Unknown return code from UAR, value is [$avp(s:uaa_return_code)]\n"); t_reply("500", "Unknown response code from UAR"); break; } } *Kamailio is getting an answer: "t_reply("500", "Error forwarding to SCSCF");"* *which means that th function I_scscf_select("0") is fine but the other function called "t_relay", is the one generating the error "Error forwarding to SCSCF".* *I-CSCF is not able to send the SIP message to S-CSCF, see this log below:* Jul 16 14:59:22 cw-aio /usr/sbin/kamailio[31604]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31599]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: scscf.epc.mnc004.mcc324.3gppnetwork.org Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches

1 0

Purpose of table location_attrs
by mayamatakeshi 16 Jul '15

16 Jul '15

Hello, I need to prepare two kamailio instances accessing the same mysql db. They will write to different location tables so i use function calls like these: Server1: save("k1_location"); lookup("k1_location"); Server2: save("k2_location"); lookup("k2_location"); This is working fine. However, checking db, I can see there is also a table called location_attrs and I am not sure what is its purpose (is this documented?). Anyway, I suspect I should also set a separate copy of it for each kamailio instance, but I am not sure how to do it. Regards, Takeshi

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