- sr-users - kamailio.org

Fw: Re: Busy Tone.. Call not coming
by Mr ankush 19 Sep '15

19 Sep '15

1 0

Dialog module isn't decreasing the profile size, and isn't entering on event routes
by José Seabra 18 Sep '15

18 Sep '15

Hi there, I'm using a dialog module in a small kamailio script that only does routing based on prefix, and relay the invite from one ipaddr to another ip addr (diferent interfaces) using $fs variavel. Now i was introducing the dialog module to limit the number of concurrent calls, and i noticed that it only can increasing the number of dialogs, and it isn't decreasing when receives the "BYE", I also noticed that when kamailio receives "BYE" the dialog event route for state "end" is not executed. To count the number of concurrent calls I'm using the following logic in main route: if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting dlg_manage(); if(get_profile_size("concurrentcalls","local","$avp(s:size)")) { if($avp(s:size) != $null) { xlog("L_INFO", "Number of calls for this server : $avp(s:size) - ID=$ci\n"); if($avp(s:size) >= 15000) { xlog("L_INFO", "Concurrent_max calls exceeded $avp(s:size) for caller '$fU' - R=$ru ID=$ci\n"); sl_send_reply("503", "Channel limit exceeded\n"); exit; } } } set_dlg_profile("concurrentcalls","local"); } Please check out the attachment with logs. Best Regards -- José Seabra

2 5

Issue with too many "listen" statements
by Sven Neuhaus 18 Sep '15

18 Sep '15

Hi, we've observed a bug with Kamailio (we are using 4.3.1): When using more than 81 "listen" statements, Kamailio starts acting up. In our case, it handles ACK packets (that follow "200 OK" messages) incorrectly: It forwards them to itself (src ip == dst ip) in a garbled form (on the loopback interface). We need more than 80 "listen" statements at the moment so I hope there's a way to increase the limit substantially. The number of file handles has been set to 32000 btw. Regards, -Sven Neuhaus

2 6

Adding user to kamailio database
by Priyaranjan Nayak 17 Sep '15

17 Sep '15

Hi All, I wanted to use kamailio with authorization. How can I add public identity and private identity to kamailio database for a user ? Thanks Priyaranjan

2 1

Hex to Dec conversion
by Yufei Tao 17 Sep '15

17 Sep '15

Hi Daniel, I've tested the patch on top of v4.2.5 and it works. Thank you very much! Yufei

1 0

Re: [SR-Users] Hex to Dec conversion
by Yufei Tao 17 Sep '15

17 Sep '15

Hi, Thanks for the replies! The first thing I looked at was the encode/decode.hexa transformations but it only transforms between ASCII characters and Hex numbers. Then I used exec_avp to call a bash command (for example, 'echo $((0x123))' gives you 291) which works. But calling something external may not be optimal I thought? I found OpenSIPS has hex2dec and dec2hex transformations and wonder if we could have these in Kamailio as well? Cheers, Yufei

2 1

Authentication issue during registration
by Austin Einter 17 Sep '15

17 Sep '15

Hi All I have a typical requirement, where I have a forwarding SBC between an endpoint and kamailio as shown below. Endpoint ----> SBC ------> Kamailio >From endpoint I am sending REGISTER request. It reaches SBC. SBC sends the REGISTER request to Kamailio, Kamailio challenges it (401), and challenge response is received at endpoint through SBC. Next endpoint adds authorization header and sends REGISTER to SBC, SBC forwards it to Kamailio. Now Kamailio again responds with 401. When we analysed Kamailio logs, we can see Kamailio log authorization response token and response token that was sent by Endpoint are same. So my question is, why Kamailio sends 401 repeatedly even if it received proper Authorization header. Is it because Kamailio received REGISTER from IP address different from endpoint. Please guide, how can I avoid this issue. Thanks Austin

3 4

add to wishlist: htable.dump type of value
by Juha Heinanen 16 Sep '15

16 Sep '15

looks like htable.dump does not tell what is the type of the value, e.g. { entry: 53 size: 1 slot: { item: { name: foo::bar value: 0 } } } -- juha

2 7

Next IRC Devel Meeting - Sep 16, 2015
by Daniel-Constantin Mierla 16 Sep '15

16 Sep '15

Hello, I would like to propose organizing the next IRC devel meeting to sync on the plans for Kamailio short term evolution. A wiki page has been created to keep track of what should be discussed there. Feel free to add your suggestions there: * http://www.kamailio.org/wiki/devel/irc-meetings/2015b First proposed date is next week on Wednesday, September 16, at 14:00 UTC (15:00 London, 16:00 Berlin, ...). The date can be changed based on availability of people willing to attend -- add your preferred date and time there. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Sep 28-30, 2015, in Berlin - http://asipto.com/u/kat

1 2

OT: IIT RTC Conference discount
by Daniel-Constantin Mierla 16 Sep '15

16 Sep '15

Hello, I have some discount code for IIT RTC Conference & Expo, taking place in Chicago, IL, USA, during October 6-8, 2015. It gives an 100USD discount, which can save a bit or make some of the available tickets completely free (like Expo Plus admission). Should someone be interested, contact me directly via email. More details about the event can be found at: * http://www.rtc-conference.com/ I will be at the event during Oct 7 and 8, presenting on the morning of Oct 8, if anyone here coming to the event or happening to be in Chicago area and want to meet and chat about Kamailio, drop me an email as well. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

assign_hf_value, hf_value_exists and exclude_hf_value broken in 4.3.2?
by Björn Bylander 15 Sep '15

15 Sep '15

Hello, 4.3.1 works well for us apart from the issue fixed by 93b297e16134b0e74cf83e3604da01355a52e700. Since 93b297e16134b0e74cf83e3604da01355a52e700 is included in 4.3.2 I tried upgrading one of our servers and now Kamailio won't even start as it apparently finds the way assign_hf_value, hf_value_exists and exclude_hf_value are used in our configuration to be incorrect. I enabled debug logging (WITH_DEBUG) and found the following: > 0(1894) DEBUG: <core> [route.c:872]: fix_actions(): fixing assign_hf_value() > 0(1894) ERROR: textopsx [textopsx.c:596]: fixup_hname_param(): ERROR: textops: cannot parse header near 'Privacy' > 0(1894) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-6) at cfg:/etc/kamailio/kamailio.cfg:642 > 0(1894) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-6) at cfg:/etc/kamailio/kamailio.cfg:644 > 0(1894) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-6) at cfg:/etc/kamailio/kamailio.cfg:661 > ERROR: error -6 while trying to fix configuration Line 642 looks like this: assign_hf_value("Privacy", "id") Commenting it out leads to the following output: > 0(2217) DEBUG: <core> [route.c:872]: fix_actions(): fixing hf_value_exists() > 0(2217) ERROR: textopsx [textopsx.c:596]: fixup_hname_param(): ERROR: textops: cannot parse header near 'Privacy' > 0(2217) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-6) at cfg:/etc/kamailio/kamailio.cfg:1016 > 0(2217) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-6) at cfg:/etc/kamailio/kamailio.cfg:1019 > 0(2217) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-6) at cfg:/etc/kamailio/kamailio.cfg:1023 > ERROR: error -6 while trying to fix configuration Line 1016 looks like this: if (hf_value_exists("Privacy", "none")) { Commenting it out leads to the following output: > 0(2292) DEBUG: <core> [route.c:872]: fix_actions(): fixing exclude_hf_value() > 0(2292) ERROR: textopsx [textopsx.c:596]: fixup_hname_param(): ERROR: textops: cannot parse header near 'Accept' > 0(2292) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-6) at cfg:/etc/kamailio/kamailio.cfg:1020 > 0(2292) ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-6) at cfg:/etc/kamailio/kamailio.cfg:1023 > ERROR: error -6 while trying to fix configuration Lines 1020 and 1021 look like this exclude_hf_value("Accept", "application/media_control+xml"); exclude_hf_value("Accept", "application/x-broadworks-call-center+xml"); Commenting them out makes Kamailio start without complaining. Btw, "kamailio -c" thinks the configuration file is fine. So, is our use of assign_hf_value, hf_value_exists and exclude_hf_value incorrect (looking at the textopsx documentation I can't see how) or was a bug introduced with 4.3.2 which causes this behaviour? Regards, Björn Bylander

2 1

Kamailio challenges repeatedly
by Austin Einter 15 Sep '15

15 Sep '15

I am trying to register with Kamailio server from my custom endpoint. I have below call flows Register -> 401 Register -> 401 When I saw Kamailio log, I do not see any error. Looks auth response is fine. Then why 401 repeatedly. Log is given blow. Please guide to fix this issue. Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: auth_db [authorize.c:208]: get_ha1(): HA1 string calculated: 8749f9da135d71526725a337008df1d5 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: auth [api.c:211]: auth_check_response(): check_response: Our result = '4aa70d676021329550b3d17e17238fac' Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: auth [api.c:218]: auth_check_response(): check_response: Authorization is OK Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 1 columns Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7f8a24d5e1d0 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7f8a24d7ceb0 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7f8a24d7cef0 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [db_res.c:54]: db_free_rows(): freeing 1 rows Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [db_row.c:97]: db_free_row(): freeing row values at 0x7f8a24d5e260 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [db_res.c:62]: db_free_rows(): freeing rows at 0x7f8a24d5e218 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7f8a24d7ce18 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: auth [challenge.c:127]: get_challenge_hf(): build_challenge_hf: realm='10.10.64.125' Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: auth [challenge.c:269]: get_challenge_hf(): auth: 'WWW-Authenticate: Digest realm="10.10.64.125", nonce="VfbImFX2x2yEeE8kEuL2IXqW2TgKu+eC"#015#012' Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: sl [sl.c:296]: send_reply(): reply in stateless mode (sl) Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(10.3.2.18, 10.3.2.18, 0) Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [tcp_main.c:2283]: tcpconn_send_put(): tcp_send: send from reader (2650 (9)), reusing fd Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [tcp_main.c:2518]: tcpconn_do_send(): tcp_send: sending... Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [tcp_main.c:2552]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f8a1c5dbe78 n=445 fd=9 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [tcp_main.c:2553]: tcpconn_do_send(): tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 10.3.2.18:6970;branch=z9hG4bK1875333638#015#012From: < sip:262099922700000@10.10.64.125;user=phone>;tag=209355482#015#012To: < sip:+499922700000@10.10.64.125;user=phone>;tag=8345fd11d20ec1ca026994bf51313cb0.a2c6#015#012Call-ID: 1340939517(a)10.10.64.125#015#012CSeq: 3 REGISTER#015#012WWW-Authenticate: Digest realm="10.10.64.125", nonce="VfbImFX2x2yEeE8kEuL2IXqW2TgKu+eC"#015#012Server: kamailio (4.2.5 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012 Sep 14 18:41:08 necs107 kamailio[2650]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Sep 14 18:41:08 necs107 kamailio[2650]: message repeated 5 times: [ DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)]

2 1

Hex to Dec conversion
by Yufei Tao 14 Sep '15

14 Sep '15

Hi, I want to convert a Hex number string into a Dec number string, e.g. Hex number "123" into Dec number "291". What is the best way? I didn't find any transformation for this. Thank you! Yufei

5 4

Very high load average after upgrade to Kamailio v4.3.2
by M S 14 Sep '15

14 Sep '15

Hi, Over last weekend i upgraded one of my test servers from Kamailio v4.3.1-4d1b65 to latest stable release v4.3.2-07690f and now kamailio goes crazy even with single call (I am using same db and configuration of course). As soon as call establishes system load average (as seen in top command) starts increasing and soon it increases beyond 6.0 and system becomes completely unresponsive, sip messages are no longer being processed by kamailio service. Even after call hangup, system remains under high load. The "htop" indicates that IO Wait time has increased substantially. Any idea what is causing this? For now i have reverted by to v4.3.1-4d1b65 but can go to v4.3.2-07690f again if you need further info or testing. Thank you.

2 6

Concurrent calls after forked invite
by Lukas Hauser 14 Sep '15

14 Sep '15

Hello, I have two UACs registered to kamailio v4.0.4 with the same URI. (bob(a)example.com) Then a third registered UAC (alice(a)example.com) sends an invite to bob(a)example.com. Both bob UACs accept the call and respond with a 200 Ok. The statefull proxy now takes the first one and cancels the second one. In my case I need both responses to be send back to alice through the proxy. I want the Invite to be forked in parallel (works) and both calls to get connected concurrently (does not work). Is there such a feature in the statefull proxy? Can this be done using a stateless proxy? I stumbled on some proposal about a "New Dialog Module Design" [1]. This is excetly what I'm looking for. Is this still just a proposal? Thanks! Lukas PS: I also posted this question in stackoverflow [2] [1] http://www.kamailio.org/dokuwiki/doku.php/modules-new-design:dialog-module-… [2] http://stackoverflow.com/questions/32519551/kamailio-concurrent-calls-after…

2 1

ha1 password authentication without plain text password
by Nabeel 14 Sep '15

14 Sep '15

Hi, My SIP client only uses the 'ha1' password column from the database to authenticate users. It does not require the 'password' column. However, I am not able to register without having the plain text password present in the 'password' column in the database. I need to change this behaviour so that user registration will occur using only the ha1 encoded password column. How can I make this change? I tried commenting out and changing the 'password_column' attribute in the config but could not get it to work.

2 1

selecting memory manager at startup
by Daniel-Constantin Mierla 14 Sep '15

14 Sep '15

Hello, (cross-posting as it is relevant to user community as well). I just pushed a bunch of patches to master branch that will allow selecting the memory manger at kamailio startup, no longer needed to re-compile sources. There are two new command line parameters: * -x name: select the memory manager for shm * -X name: select the memory manager for pkg * if -x is provided but -X is not provided, then pkg uses same memory manager as shm * name can be: * fm - fast malloc (f_malloc) * qm - quick malloc (q_malloc) * default memory manager is fm (what was default so far) Both f_malloc and q_malloc are now compiled, selecting one or the other being possible. Other managers expected to be added in the list soon, especially TLSF, added not that long ago by Camille Oudot. Maybe reviving Doug Lea malloc (existing in dl_malloc.c) worth a try now that can be easily used. The debugging mode for each memory manager is controlled by DB_SR_MEMORY define via Makefile.defs -- right now it is enabled. All enabled memory mangers must be compiled either with or without DB_SR_MEMORY (it won't work to have one memory manger with DB_SR_MEMORY and another one without). Using system malloc for private (pkg) memory is possible by removing the define of PKG_MALLOC from Makefile.defs (same as it was before). This is a mode suitable for troubleshooting using external tools such as valgrind, because it does inline replacement of pkg_malloc() with malloc(), so those tools can track the place of allocation. I plan to add a system malloc custom manager, to allow selecting it with -X (system malloc won't work for shm management, because it is only for private memory). There is no performance impact at runtime, it is only the startup procedure that replaces the pointers to memory management functions. Hopefully with the new mechanism will be able to track easier memory issues -- e.g., qm malloc is better for troubleshooting, using it no longer requires recompilation, just starting kamailio with -x qm. Feedback, especially testing, will be very appreciated! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Sep 28-30, 2015, in Berlin - http://asipto.com/u/kat

1 0

no response, re-route calls to a different address
by Al S 14 Sep '15

14 Sep '15

Hi, Can we make Kamailio to re-route calls from an address that does not respond to a different address? Thanks,Al

3 4

Kamailio v4.3.2 Released
by Daniel-Constantin Mierla 10 Sep '15

10 Sep '15

Hello, Kamailio SIP Server v4.3.2 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.2. For more details about version 4.3.2 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/09/kamailio-v4-3-2-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

Problem with registering softphones
by rasx0＠Safe-mail.net 10 Sep '15

10 Sep '15

Hello! New user here. I'm testing Kamailio installation in testing environment. SIP domain is reachable, Kamailio itself is working fine yet I have a problem with registering softphones. I've tried 3 sip apps and with only one I am able to register with server. Server itself is "vanilla", no changes made. There is user, domain, pwd defined and nothing else. App that successfully connects is X-Lite, after unchecking "Register with domain and receive calls" and checking "Set outbound as domain". Other apps are: Zoiper, Jitsi. After many "checks" and "unchecks" of literary every option I still can't get them to work with me. I'm looking for any kind of hints as of how to get it to register as well as where may I find kamailio logs?

2 1

Releasing Kamailio v4.3.2
by Daniel-Constantin Mierla 10 Sep '15

10 Sep '15

Hello, I am considering to release a new version from branch 4.3 later this week, on Thursday, September 10. If there are issues not yet reported, then use github tracker to do it: - 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 Kamailio Advanced Training, Sep 28-30, 2015, in Berlin - http://asipto.com/u/kat

2 7

No SDP offer answer -> therefore we can not do Rx AAR 1(875) ERROR
by Jan Devai 10 Sep '15

10 Sep '15

Hi Guys, can anybody help us find why we do not get SDP offer answer, see error below: dev_pcscf: 3(879) DEBUG: ims_qos [mod.c:643]: w_rx_aar(): No SDP offer answer -> therefore we can not do Rx AAR 1(875) ERROR: <script>: ACK ( sip:+4620950160102@ims.mnc004.mcc232.3gppnetwork.org <sip:+421950160102@ims.mnc003.mcc231.3gppnetwork.org> (10.50.21.133:5080) to sip:+4620950160101@ims.mnc004.mcc232.3gppnetwork.org;user=phone <sip:+421950160101@ims.mnc003.mcc231.3gppnetwork.org;user=phone>, 3FD60673-55F09407000575C3-18269700) Since we do not get the information the SubscriptionID should be imsi(a)mncxx.mccxx.3gppnetwork.org in our AAR and instead it is contructed as imsi@ip:port which creates a REJECTION in AAA. Any help about where to correct this fault, or where exactly in the code this reception happens and/or anything will be appreciated BIG TIME :-) Thank you Jan

2 1

MEM_JOIN_FREE
by Uri Shacked 09 Sep '15

09 Sep '15

Hi, Does MEM_JOIN_FREE is compiled by default in kamailio 4.X.X ? thanks, Uri

4 6

Kamailio systemd scripts
by Daniel-Constantin Mierla 09 Sep '15

09 Sep '15

Hello, anyone here familiar with systemd that can provide the scripts for kamailio? Systemd is default now on last debian, so would be good to have them as well as I want to close the issue 294: - https://github.com/kamailio/kamailio/issues/294 Any help will be appreciated. Follow up on this list or on the github issue tracker. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Sep 28-30, 2015, in Berlin - http://asipto.com/u/kat

4 5

Dialog module issue
by Yuriy Gorlichenko 09 Sep '15

09 Sep '15

Hello. I try to manage by dialog module every signaling session, that goes through my proxy. I added newx mod params modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_flag", 4) modparam("dialog", "initial_cbs_inscript", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30 And at request route I added this for all invite methods if (is_method("INVITE")){ $dlg_ctx(timeout_route) = "DIALOG_END"; $avp(i:10)=43200; $dlg_ctx(timeout_bye) = 1; dlg_set_property("ka-src"); dlg_set_property("ka-dst"); dlg_manage(); xlog("L_INFO","Dialog manage is {$ct}\n"); ... } So after this As I think every session must be dialog managed session and every leg of call must be checked by keepalive OPTIONS packets but no one OPTIONS request generated wen session goes through my proxy. thanks

1 0

Using PSTN as "fallback"
by Michael Nielsen 09 Sep '15

09 Sep '15

I have Kamailio running and connected to a PSTN gateway. My subscribers are named ex. +442071234567 - same as their real GSM number from my PSTN gateway. I'm using the standard kamailio.cfg which ships with version 4.3. When I'm trying to dial SIP client to SIP client I would like to have Kamailio route the call internally if a subscriber exists with ex. +442071234567. If no subscriber exists with ex. +442071234567 it should send it to my PSTN gateway. As it is now it seems as if it are trying to both call "internally" and via the PSTN gateway. How should one fix this issue the best way?

3 5

Kamailio crash [receive_fd: EOF on 19]
by SamyGo 09 Sep '15

09 Sep '15

Hi, I'm randomly getting crash in my Kamailio with an error in log files like this: <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 19 ALERT: <core> [main.c:775]: handle_sigs(): child process 6853 exited by a signal 11 ALERT: <core> [main.c:778]: handle_sigs(): core was generated INFO: <core> [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD *Version of Kamailio is:* [root@kamailio /]#kamailio -V version: kamailio 4.1.4 (x86_64/linux) 39adca flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 39adca compiled on 03:48:50 Aug 1 2014 with gcc 4.4.7 And gdb bt output is attached in text file. One thing out that is common in all the previous core files is this: #12 0x00000000004a54ab in receive_msg ( buf=0x924600 "INVITE sip:+17036833500@14.131.165.9:5073 SIP/2.0\r\nVia: SIP/2.0/UDP 14.55.2.43:5060;branch=z9hG4bK0eB8f68591a19b9b8f0\r\nFrom: \"Anonymous\" <sip:Anonymous@Anonymous.invalid>;tag=gK0e13e132\r\nTo: <sip:+170"..., len=1018, rcv_info=0x7fff3dbee970) at receive.c:212 #13 0x000000000053c9a8 in udp_rcv_loop () at udp_server.c:536 Does this mean that Kamailio can't understand the From Domain: Anonymous.invalid and hence crashes ? Thanks, Sammy

2 6

Best practise for limiting concurrent calls across a cluster
by jay binks 08 Sep '15

08 Sep '15

Hey all, So I have a cluster of Kamailio servers ( 4 servers currently, soon to be 8 ), I'm looking for suggestions about the BEST way to achieve concurrent call limiting on a per customer basis, across the whole cluster. Initially I mis-read the dialog module documentation and assumed that dialog would provide me this ability, when used with a database. however it seems that the dialog module does not pull data from the DB after the initial startup. I know I can use sql ops to increment and decrement using event_route[dialog:start] and event_route[dialog:end]. however the database I've chosen ( for other valid reasons ) does not have an atomic increment and decrement. I could add yet another DB, but that just adds more failure points. so lets forget my setup, Im wanting suggestions about the BEST setup for this sort of thing. while remaining fault tolerant, and preferably without relying on any single point of failure. Sincerely Jay

7 13

core parameter onsend_route_reply=1 doesn't execute onsend_route block for received replies
by Julia Boudniatsky 07 Sep '15

07 Sep '15

Hello, I set some logs in onsend_route and it works fine for requests. With setting core parameter onsend_route_reply=1 ( yes, on), i expect to see logs for replays also, but it not occurs. Is this a bug or I miss something? version: kamailio 4.2.6 (x86_64/linux) db77ac BR, Julia

2 4

Siremis 4.2 with PHP 5.6
by Patrick Bezensiertblubb 07 Sep '15

07 Sep '15

2 2

Kamailio + Rtpengine < - > Freeswitch + ZRTP
by Alexandru Covalschi 07 Sep '15

07 Sep '15

Hello! Was trying to perfrom subj interconnection via http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbс <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> via that guide. Kamailio and FS are on same host on Amazon EC2. The issue is - when I have proxy_media=true on FS, ZRTP is OK, but the leg_b user can't hear anything (but his voice is transmitted OK). When I set proxy-media to false the voice is OK, but something is with ZRTP so Jitsi can't establish ZRTP handshake. When I set bypass-media=true on FS both ZRTP and voice established, but quality is sometimes bad and the delay is too big. Anyone has any ideas? Kamailio 4.3, FS 1.4 <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 1

Thoughts on setup with Kamailio and FreeSWITCH
by Michael Nielsen 07 Sep '15

07 Sep '15

Hello everyone... I'm testing various ways of setting up Kamailio and FreeSWITCH (as VAS-handler). I'm struggling a bit on what Kamailio should do and FreeSWITCH should do. My basic setup is to handle voicemails, internal calls between subscribers and outgoing calls to a SIP gateway to GSM. My idea is to have Kamailio handle all calls and routing between subscribers and the SIP gateway and only have FreeSWITCH handle voicemail (to save the audio file and e-mail it). How does that sound? Or should I route everything to FreeSWITCH via Kamailio and have FreeSWITCH route calls between subcsribers and SIP gateway as well? What are best practises etc...?

2 1

Question about in call both side redirection
by Андрей Ярин 07 Sep '15

07 Sep '15

Hello (sorry for my bad english) - i try to create voice record service by request. User A call to user B. In call by pressing combination like *55 Kamailio must redirect both sides to asterisk, whitch create dynamic conference room with recording. As i understand i need to use dlg_refer() from dialog module, but in log file i get: Konsole output Sep 4 10:45:14 voipc-node2 /usr/local/sbin/kamailio[18941]: ERROR: dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available Sep 4 10:45:14 voipc-node2 /usr/local/sbin/kamailio[18941]: ERROR: dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t In script i try to refer with: dlg_refer("callee", "sip:100@10.10.9.209"); dlg_refer("caller", "sip:100@10.10.9.209"); -- Ярин Андрей Инженер ООО "И.С.-Телеком" 8(351)7786878 доб 113 8(351)2555724 доб 113 8(351)2555787 доб 113

2 1

Route all traffic from Kamailio to FreeSWITCH after inital checks...
by Michael Nielsen 07 Sep '15

07 Sep '15

I'm running the standard Debian 7 Kamailio 4.3 package. I've added the following to my kamailio.cfg: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS #!define WITH_ANTIFLOOD After authentication, nat etc. is done is there an easy way to transfer everything else to FreeSWITCH? I've looked at the dispatcher module, but don't know where to put the "transfer" in kamailio.cfg?

3 4

Re: [SR-Users] webrtc
by Slava Bendersky 07 Sep '15

07 Sep '15

Hello Giovanni, I fixed webrtc call working right now. I checked debug again and I so 408 error. There was missing record_route() for first INVITE. Thank you for help. Slava. From: "Giovanni Maruzzelli" <gmaruzz(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 5 September, 2015 07:50:29 Subject: Re: [SR-Users] webrtc First thing, check that you can have a webRTC call directly with FreeSWITCH, without passing by Kamailio. If that works, and it does not work through Kamailio, then provide kamailio config, freeswitch config, and debug log from both. On Sat, Sep 5, 2015 at 12:51 AM, Slava Bendersky < volga629(a)skillsearch.ca > wrote: Hello Everyone, I am trying setup webrtc call with freeswitch. [root@canlvprx01 kamailio]# rpm -qa | grep kamailio kamailio-4.3.1-4.4.fc21.x86_64 Case: INTERNET eth0 | eth1 LAN | Web Browser ---->webrtc ----> kamailio ----> freeswitch Getting this error Sep 4 00:22:06 canlvprx01 /usr/sbin/kamailio[11820]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:10 canlvprx01 /usr/sbin/kamailio[11817]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:14 canlvprx01 /usr/sbin/kamailio[11818]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Tried check debug don't see any thing special. Any help thank you. Please let me know which logs will good to provide. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] webrtc
by Slava Bendersky 06 Sep '15

06 Sep '15

Hello Giovanni, Here SDP webrtc call traffic capture on kamailio side http://fpaste.org/264289/15798761/ password from paste: kamailio Slava. From: "Giovanni Maruzzelli" <gmaruzz(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 5 September, 2015 07:50:29 Subject: Re: [SR-Users] webrtc First thing, check that you can have a webRTC call directly with FreeSWITCH, without passing by Kamailio. If that works, and it does not work through Kamailio, then provide kamailio config, freeswitch config, and debug log from both. On Sat, Sep 5, 2015 at 12:51 AM, Slava Bendersky < volga629(a)skillsearch.ca > wrote: Hello Everyone, I am trying setup webrtc call with freeswitch. [root@canlvprx01 kamailio]# rpm -qa | grep kamailio kamailio-4.3.1-4.4.fc21.x86_64 Case: INTERNET eth0 | eth1 LAN | Web Browser ---->webrtc ----> kamailio ----> freeswitch Getting this error Sep 4 00:22:06 canlvprx01 /usr/sbin/kamailio[11820]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:10 canlvprx01 /usr/sbin/kamailio[11817]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:14 canlvprx01 /usr/sbin/kamailio[11818]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Tried check debug don't see any thing special. Any help thank you. Please let me know which logs will good to provide. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Topoh configuration question
by Asgaroth 06 Sep '15

06 Sep '15

Hi all, I have a quick question with regards the topoh module, is it possible to have the topoh module only operate on a particular interface and direction. I have some edge proxies and I am trying out the topoh module, what I see happen is that messages originating from behind/inside the edge proxy is topoh'd out, which is exactly what I was looking for, but it appears to do the same for messages originating from the wild, so messages pushed in behind/inside the proxy is also encoded, which is not what I was expecting. Is it possible to configure the topoh module to activate for messages being sent out of the edge proxy into the "wild", but not the other way around, or, decode if it finds an encoded messages who's destination is "behind" the proxy. Any thoughts are much appreciated. Thanks

2 1

webrtc
by Slava Bendersky 05 Sep '15

05 Sep '15

Hello Everyone, I am trying setup webrtc call with freeswitch. [root@canlvprx01 kamailio]# rpm -qa | grep kamailio kamailio-4.3.1-4.4.fc21.x86_64 Case: INTERNET eth0 | eth1 LAN | Web Browser ---->webrtc ----> kamailio ----> freeswitch Getting this error Sep 4 00:22:06 canlvprx01 /usr/sbin/kamailio[11820]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:10 canlvprx01 /usr/sbin/kamailio[11817]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:14 canlvprx01 /usr/sbin/kamailio[11818]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Tried check debug don't see any thing special. Any help thank you. Please let me know which logs will good to provide. Slava.

2 1

Re: [SR-Users] CRITICAL log on kamailio startup
by Nuno Reis 04 Sep '15

04 Sep '15

Good morning. Has anyone seen this too? Looking forward to hear from you. BR, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/> On Fri, Aug 14, 2015 at 1:41 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: > Hi guys. > > I've noticed this CRITICAL log line on kamailio startup: > > CRITICAL: dialog [dlg_cb.c:137]: register_dlgcb(): DLGCB_CREATED type > registered after shutdown!?! > > > This line started to show up after upgrade from 4.2.x to 4.3 and now to > 4.3.1. > Everything seems to be working fine and a really don't see any harm > anywhere. > Could you please comment on this? > > Cheers, > > -- > > *Nuno Miguel Reis* | *Unified Communication** Systems* > M. +351 913907481 | nreis(a)wavecom.pt > WAVECOM-Soluções Rádio, S.A. > Cacia Park | Rua do Progresso, Lote 15 > 3800-639 AVEIRO | Portugal > T. +351 309 700 225 | F. +351 234 919 191 > *GPS > <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> > | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* > > <http://www.wavecom.pt/> > <http://www.wavecom.pt/pt/> > > >

2 2

Sounds works differently on different clients.
by Michael Nielsen 04 Sep '15

04 Sep '15

This is my setup: Kamailio -> FreeSWITCH (for voicemail) ...SIP gateway for connecting GSM So every call is routed directly via Kamailio - between SIP clients and in/out from the GSM space. Testing with a cellular phone as one client and X-Lite on Mac OS X as the other - everything works. Testing with a cellular phone as one client and Bria on iOS as the other - everything works except there's no sounds coming into Bria. The person on the cellular can hear everything (sound coming from Bria), but Bria client cannot hear sound from cellular phone. Is it something to do with codec or could it be something else?

2 3

408 Request Timeout
by vamsi 04 Sep '15

04 Sep '15

I am an web developer and want to create a WEBRTC application. My knowledge level in SIP is beginner. I want my application to talk to kamailio and in process of setting up kamailio by following the below articles. http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket http://kamailio.org/docs/modules/4.3.x/modules/websocket.html I was able to connect (using try.jssp.net) to kamailio as multiple clients for initial setup and successfully able to get voice and video working. But I'm unable to send messages between clients and receive "408 Request Timeout" error. I have the presence and presence_xml modules loaded and have the # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") # ----- ctl params ----- modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") Below is the log. Any help is appreciated. Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[AUTH] c=[/usr/local/etc/kamailio/kamailio.cfg] l=825 a=2 n=exit Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [receive.c:278]: receive_msg(): cleaning up Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [tcp_read.c:1153]: ws_process_msg(): WebSocket Message: [[>>>#012??#002?E?<<<]] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_conn.c:429]: wsconn_get(): wsconn_get for id [2] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_conn.c:437]: wsconn_get(): wsconn_get returns wsc [0x7f06a2a76298] refcnt [2] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_frame.c:415]: decode_and_validate_ws_frame(): decoding WebSocket frame Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_frame.c:456]: decode_and_validate_ws_frame(): supported non-control frame: 0x1 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_frame.c:555]: decode_and_validate_ws_frame(): Rx (decoded): MESSAGE sip:abc123@ip-205-64-10.dqa.capitalone.com SIP/2.0#015#012Via: SIP/2.0/WS kjh0uotuqma6.invalid;branch=z9hG4bK9552000#015#012Max-Forwards: 69#015#012To: <sip:abc123@ip-205-64-10.dqa.capitalone.com>#015#012From: "xyz987" <sip:xyz987@ip-205-64-10.test.com:8088>;tag=c1joq54vne#015#012Call-ID: orel8as144p4ebuuf96n#015#012CSeq: 1977 MESSAGE#015#012Proxy-Authorization: Digest algorithm=MD5, username="xyz987", realm="ip-205-64-10.test.com", nonce="Vee3iFXntlzqK/kDTxUAIxwI55RzH/ll", uri=" sip:abc123@ip-205-64-10.test.com", response="0d513943ed7a0234b7016c5f4546b0b7"#015#012Content-Type: text/plain#015#012Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER#015#012Supported: outbound#015#012User-Agent: JsSIP 0.7.4#015#012Content-Length: 13#015#012#015#012samplemsgwork Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_frame.c:659]: ws_frame_receive(): Rx SIP message:#012MESSAGE sip:abc123@ip-10-255-54-20.test.com SIP/2.0#015#012Via: SIP/2.0/WS kjh0uotuqma6.invalid;branch=z9hG4bK9552000#015#012Max-Forwards: 69#015#012To: <sip:abc123@ip-10-255-54-20.test.com>#015#012From: "xyz987" < sip:xyz987@ip-10-255-54-20.test.com:8088>;tag=c1joq54vne#015#012Call-ID: orel8as144p4ebuuf96n#015#012CSeq: 1977 MESSAGE#015#012Proxy-Authorization: Digest algorithm=MD5, username="xyz987", realm="ip-10-255-54-20.test.com", nonce="Vee3iFXntlzqK/kDTxUAIxwI55RzH/ll", uri=" sip:abc123@ip-10-255-54-20.test.com", response="0d513943ed7a0234b7016c5f4546b0b7"#015#012Content-Type: text/plain#015#012Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER#015#012Supported: outbound#015#012User-Agent: JsSIP 0.7.4#015#012Content-Length: 13#015#012#015#012samplemsgwork Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <MESSAGE> Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: < sip:abc123@ip-10-255-54-20.test.com> Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK9552000>; state=16 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [receive.c:134]: receive_msg(): After parse_msg... Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [receive.c:177]: receive_msg(): preparing to run routing scripts... Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=506 a=5 n=route Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=639 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=634 a=25 n=mf_process_maxfwd_header Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 69 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=644 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=639 a=25 n=is_method Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=648 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=644 a=26 n=sanity_check Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [46]; uri=[sip:abc123@ip-10-255-54-20.test.com] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [< sip:abc123@ip-10-255-54-20.test.com>#015#012] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1977> <MESSAGE> Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=13 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=c1joq54vne Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=529 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=509 a=25 n=nat_uac_test Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=514 a=55 n=force_rport Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=524 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=515 a=25 n=is_method Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=524 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=519 a=24 n=add_contact_alias Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: nathelper [nathelper.c:958]: add_contact_alias_0_f(): no contact header Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=529 a=5 n=route Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=846 a=55 n=force_rport Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=857 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=847 a=25 n=nat_uac_test Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=853 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=848 a=25 n=is_method Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=853 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=851 a=24 n=is_first_hop Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: siputils [sipops.c:159]: w_is_first_hop(): no 2nd via found - first hop Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=852 a=24 n=set_contact_alias Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=854 a=39 n=setflag Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[NATDETECT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=857 a=2 n=return Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=532 a=25 n=is_method Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=5 n=route Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[WITHINDLG] c=[/usr/local/etc/kamailio/kamailio.cfg] l=654 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[WITHINDLG] c=[/usr/local/etc/kamailio/kamailio.cfg] l=652 a=24 n=has_totag Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: siputils [checks.c:97]: has_totag(): no totag Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[WITHINDLG] c=[/usr/local/etc/kamailio/kamailio.cfg] l=652 a=2 n=return Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=549 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=545 a=24 n=t_precheck_trans Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=549 a=24 n=t_check_trans Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=4 global id=3 T start=(nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=65389, isACK=0 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=4 global id=4 T end=(nil) Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=552 a=5 n=route Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[AUTH] c=[/usr/local/etc/kamailio/kamailio.cfg] l=833 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[AUTH] c=[/usr/local/etc/kamailio/kamailio.cfg] l=820 a=25 n=is_method Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 31==12 && [ ip-10-255-54-20.test.com] == [10.205.64.10] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 8088 (advertise 0) matches port 8088 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 31==12 && [ ip-10-255-54-20.test.com] == [10.205.64.10] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 8088 (advertise 0) matches port 8088 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 31==12 && [ ip-10-255-54-20.test.com] == [10.205.64.10] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 8080 (advertise 0) matches port 8088 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 31==12 && [ ip-10-255-54-20.test.com] == [10.205.64.10] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5061 (advertise 0) matches port 8088 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 31==12 && [ ip-10-255-54-20.test.com] == [10.205.64.10] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 4443 (advertise 0) matches port 8088 Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[AUTH] c=[/usr/local/etc/kamailio/kamailio.cfg] l=828 a=16 n=if Sep 2 22:54:20 ip-10-255-54-20 kamailio[10177]: exec: *** cfgtrace:request_route=[AUTH] c=[/usr/local/etc/kamailio/kamailio.cfg] l=823 a=27 n=auth_check Sep 2 22:54:20 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:457]: wsconn_get_list(): wsconn_get_list Sep 2 22:54:20 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a775a8] prev => [(nil)] next => [0x7f06a2a76298] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a76298] prev => [0x7f06a2a775a8] next => [(nil)] Sep 2 22:54:20 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a775a8] id [1] ref++ Sep 2 22:54:20 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a76298] id [2] ref++ Sep 2 22:54:25 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:501]: wsconn_get_list(): wsconn_get_list returns list [0x7f06ab4f8350] with [2] members Sep 2 22:54:25 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:511]: wsconn_put_list(): wsconn_put_list [0x7f06ab4f8350] Sep 2 22:54:25 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a775a8] refcnt [2] Sep 2 22:54:25 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a775a8] refcnt [1] Sep 2 22:54:25 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:25 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1] Sep 2 22:54:26 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa40820, 9, -1, 0x10) fd_no=2 called Sep 2 22:54:26 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [tcp_read.c:1450]: release_tcpconn(): releasing con 0x7f06a2a8c3b8, state 1, fd=9, id=2 Sep 2 22:54:26 ip-10-255-54-20 kamailio[10177]: DEBUG: <core> [tcp_read.c:1451]: release_tcpconn(): extra_data (nil) Sep 2 22:54:26 ip-10-255-54-20 kamailio[10178]: DEBUG: <core> [tcp_main.c:3243]: handle_tcp_child(): reader response= 7f06a2a8c3b8, 1 from 3 Sep 2 22:54:26 ip-10-255-54-20 kamailio[10178]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0x9fc580, 34, 2, 0x7f06a2a8c3b8), fd_no=23 Sep 2 22:54:26 ip-10-255-54-20 kamailio[10178]: DEBUG: <core> [tcp_main.c:3370]: handle_tcp_child(): CONN_RELEASE 0x7f06a2a8c3b8 refcnt= 1 Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:457]: wsconn_get_list(): wsconn_get_list Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a775a8] prev => [(nil)] next => [0x7f06a2a76298] Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a76298] prev => [0x7f06a2a775a8] next => [(nil)] Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a775a8] id [1] ref++ Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a76298] id [2] ref++ Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:501]: wsconn_get_list(): wsconn_get_list returns list [0x7f06ab4f8350] with [2] members Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:511]: wsconn_put_list(): wsconn_put_list [0x7f06ab4f8350] Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a775a8] refcnt [2] Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a775a8] refcnt [1] Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:26 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1] Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:457]: wsconn_get_list(): wsconn_get_list Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a775a8] prev => [(nil)] next => [0x7f06a2a76298] Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a76298] prev => [0x7f06a2a775a8] next => [(nil)] Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a775a8] id [1] ref++ Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a76298] id [2] ref++ Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:501]: wsconn_get_list(): wsconn_get_list returns list [0x7f06ab4f8350] with [2] members Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:511]: wsconn_put_list(): wsconn_put_list [0x7f06ab4f8350] Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a775a8] refcnt [2] Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a775a8] refcnt [1] Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:27 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1] Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:457]: wsconn_get_list(): wsconn_get_list Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a775a8] prev => [(nil)] next => [0x7f06a2a76298] Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a76298] prev => [0x7f06a2a775a8] next => [(nil)] Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a775a8] id [1] ref++ Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a76298] id [2] ref++ Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:501]: wsconn_get_list(): wsconn_get_list returns list [0x7f06ab4f8350] with [2] members Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:511]: wsconn_put_list(): wsconn_put_list [0x7f06ab4f8350] Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a775a8] refcnt [2] Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a775a8] refcnt [1] Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:28 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1] Sep 2 22:54:29 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:457]: wsconn_get_list(): wsconn_get_list Sep 2 22:54:29 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a775a8] prev => [(nil)] next => [0x7f06a2a76298] Sep 2 22:54:29 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a76298] prev => [0x7f06a2a775a8] next => [(nil)] Sep 2 22:54:29 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a775a8] id [1] ref++ Sep 2 22:54:29 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a76298] id [2] ref++ Sep 2 22:54:31 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:501]: wsconn_get_list(): wsconn_get_list returns list [0x7f06ab4f8350] with [2] members Sep 2 22:54:31 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:511]: wsconn_put_list(): wsconn_put_list [0x7f06ab4f8350] Sep 2 22:54:31 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a775a8] refcnt [2] Sep 2 22:54:31 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a775a8] refcnt [1] Sep 2 22:54:31 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:31 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1] Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: tm [t_reply.c:1230]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: tm [t_reply.c:1748]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: <core> [msg_translator.c:158]: check_via_address(): (172.18.116.115, kjh0uotuqma6.invalid, 0) Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: <core> [mem/shm_mem.c:101]: _shm_resize(): WARNING:vqm_resize: resize(0) called Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: websocket [ws_conn.c:429]: wsconn_get(): wsconn_get for id [2] Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: websocket [ws_conn.c:437]: wsconn_get(): wsconn_get returns wsc [0x7f06a2a76298] refcnt [2] Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: websocket [ws_frame.c:751]: ws_frame_transmit(): Tx message:#012SIP/2.0 408 Request Timeout#015#012Via: SIP/2.0/WS kjh0uotuqma6.invalid;branch=z9hG4bK2462101;rport=59540;received=172.18.116.115#015#012To: <sip:abc123@ip-10-255-54-20.test.com>;tag=b44d120c617c843b219e52eae8b5e9ff-3dde#015#012From: "xyz987" <sip:xyz987@ip-10-255-54-20.test.com:8088>;tag=5qg1sitf3i#015#012Call-ID: orel8dhn00r81cega00f#015#012CSeq: 862 MESSAGE#015#012Server: kamailio (4.4.0-dev2 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012 Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: websocket [ws_frame.c:150]: encode_and_send_ws_frame(): encoding WebSocket frame Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: websocket [ws_frame.c:181]: encode_and_send_ws_frame(): supported non-control frame: 0x1 Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: <core> [tcp_main.c:2211]: tcpconn_send_put(): tcp connection found (0x7f06a2a8c3b8), acquiring fd Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: <core> [tcp_main.c:2221]: tcpconn_send_put(): c=0x7f06a2a8c3b8, n=16 Sep 2 22:54:31 ip-10-255-54-20 kamailio[10178]: DEBUG: <core> [tcp_main.c:3476]: handle_ser_child(): read response= 7f06a2a8c3b8, 2, fd -1 from 5 (10164) Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: <core> [tcp_main.c:2251]: tcpconn_send_put(): after receive_fd: c= 0x7f06a2a8c3b8 n=8 fd=8 Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7f06a2a8c3b8 n=419 fd=8 Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf=#012?~#001?SIP/2.0 408 Request Timeout#015#012Via: SIP/2.0/WS kjh0uotuqma6.invalid;branch=z9hG4bK2462101;rport=59540;received=172.18.116.115#015#012To: <sip:abc123@ip-10-255-54-20.test.com>;tag=b44d120c617c843b219e52eae8b5e9ff-3dde#015#012From: "xyz987" <sip:xyz987@ip-10-255-54-20.test.com:8088>;tag=5qg1sitf3i#015#012Call-ID: orel8dhn00r81cega00f#015#012CSeq: 862 MESSAGE#015#012Server: kamailio (4.4.0-dev2 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012 Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:31 ip-10-255-54-20 kamailio[10164]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1] Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:457]: wsconn_get_list(): wsconn_get_list Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a775a8] prev => [(nil)] next => [0x7f06a2a76298] Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:465]: wsconn_get_list(): counter wsc [0x7f06a2a76298] prev => [0x7f06a2a775a8] next => [(nil)] Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a775a8] id [1] ref++ Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:492]: wsconn_get_list(): wsc [0x7f06a2a76298] id [2] ref++ Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:501]: wsconn_get_list(): wsconn_get_list returns list [0x7f06ab4f8350] with [2] members Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:511]: wsconn_put_list(): wsconn_put_list [0x7f06ab4f8350] Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a775a8] refcnt [2] Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a775a8] refcnt [1] Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f06a2a76298] refcnt [2] Sep 2 22:54:32 ip-10-255-54-20 kamailio[10173]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f06a2a76298] refcnt [1]

2 2

Kamailio memmory leak after it does not receive response for CANCEL message
by Bui Minh Yen 03 Sep '15

03 Sep '15

Hi everyone, I am using Kamailio-4.1.5 and Kamailio-4.2.5 but with this both version I had a problem: Connection diagrams: SIP client ---- SBC --- Kamailio --- MGC/MGW --- GSM/PSTN subscriber Scenario: - Caller (SIP) make a new call to GSM (PSTN) subscriber - When GSM/PSTN is ringing or not ringing, Caller cancel call (Caller send CANCEL message) - But MGC does not send response message for CANCEL message - After about 120s (fr_inv_timer), Kamailio send 487 Request Terminated to SBC. >From this time, Kamailio have memory leak (tm does not free transactions) I reproduced this bugs 4-5 times and same problem is happen I attached capture file and log when I show shared memory (using kamcmd core.shmmem) and tm statistics (using kamcmd tm.stats). And time Kamailio start memory leak is 08-30-15-10:18:53 Could you help me solve this problem? Thank you so much! -- --------------------------- *Bui Thi Minh Yen*

2 2

A little offtop - a question about Kamailio Admin Book
by Alexandru Covalschi 03 Sep '15

03 Sep '15

Just wanted to ask when will http://www.asipto.com/sw/kamailio-admin-book/ become available in final version? Sorry for offtop, didn't know where to ask about that :) -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 1

registered4() is not adding xavp with details of the record
by Andrew Pogrebennyk 03 Sep '15

03 Sep '15

Hi, I was trying to use registered("location", "$ru", 0, 1) Last parameter is the flag according to http://kamailio.org/docs/modules/stable/modules/registrar.html#registrar.f.… flag values is as follows: 1 - set xavp_rcd with value from matched contact But I'm getting NULL instead of ruid.. While the same works after lookup("location"). So I took a quick look into the code and that confirms that registered4 does not add the xavp with details of the record (ruid), i.e. it does not do what the lookup_helper does. Is this done on purpose or an oversight? While fixing this it might be reasonable to introduce a new function for setting the XAVPs and call it from lookup and registered4 functions, especially since we are going to extend the attributes list beyond just ruid, but right now I'm struggling just to understanding how the XAVP should be built.. Ideas? Thanks, Andrew

2 1

during registration nonce expired, after backwards time shift
by Vasiliy Ganchev 03 Sep '15

03 Sep '15

Hi list! (sorry for sending first mail to wrong sub forum) Have a following issue: Server with Kamailio restart with wrong time (why - this is separate part of my investigations), timeshift e.g. for 2 hours in feature. After ntp daemon adjust correct time (move time backwards), Kamailio do not accept REGISTERs, pv_www_authenticate answer with code -4 (nonce expired) I've reade the description of: http://kamailio.org/docs/modules/4.2.x/modules/auth.html#auth.p.nonce_expire and http://kamailio.org/docs/modules/4.2.x/modules/auth.html#auth.p.nonce_auth_… This descriptions mention situation with backwards timeshift from future. As I understand, Kamailio do not authorise the registration, whose nonces where generated in future (before ntp shifted the time). But in my tests, Kamailio do not accept even REGISTER from UA, sent after time shift (with new nonces, that as I expected has to use correct time for nonce generation, and be authorised with no problem). Only Kamailio process restart help to cope with the issue. It looks like for nonce generating Kamailio use old time (in feature, that was before timeshift). I wander, does the Kamailio behave in described case - as is expected? If I am wrong in some of my assumptions, please point out. version: kamailio 4.2.5 Thank in advance! Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/during-registration-nonce-expired-a… Sent from the Users mailing list archive at Nabble.com.

2 12

[Kamailio 4.2] Crash during burn-in test
by Jack Wang 03 Sep '15

03 Sep '15

[Steps] 1. one MCU call multi-endpoints 2. MCU hangup all endpoints repeatedly. And all through Kamailio proxy. [Results] For a while, Kamailio crashed. Followings are related logs: ========================================== Sep 2 19:08:34 ./kamailio[3712]: : tm [t_fwd.c:1632]: t_send_branch(): BUG: t_send_branch: retr. already started for 0x41848fa0 Sep 2 19:08:34 ./kamailio[3712]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) At this point the related behavior is that Kamailio received two INFO requests simultaneously. I wonder whether this is the cause. And maybe it's related to the later crash? Sep 2 19:08:35 ./kamailio[3713]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 11 Sep 2 19:08:35 ./kamailio[3703]: ALERT: <core> [main.c:784]: handle_sigs(): child process 3704 exited by a signal 11 Sep 2 19:08:35 ./kamailio[3703]: ALERT: <core> [main.c:787]: handle_sigs(): core was generated Sep 2 19:08:35 ./kamailio[3703]: : <core> [mem/f_malloc.c:586]: fm_free(): BUG: fm_free: bad pointer 0x7239b6c4 (out of memory block!), called from tm: h_table.c: free_cell(157) - aborting Any suggestions? Thanks. ==========================================

2 1

Change caller id with Kamailio
by Michael Nielsen 02 Sep '15

02 Sep '15

The same way I can change called party id with $rU, is there also a way to change the caller id of the person making the call?

2 2

Add country code on incoming calls - if not present
by Michael Nielsen 02 Sep '15

02 Sep '15

I've added a custom column in my subscriber table in MySQL - called country_code. This contains +44 etc. for subscribers. Is it possible to, if a subscriber has not entered either + or 00, to append the value from country_code of the specific subscriber to the dialled number? And if the user has dialled 00, to replace it with +.

3 7

Kamailio TLS handshake fails above 6500 TLS connections
by Priyaranjan Nayak 02 Sep '15

02 Sep '15

Hi All, We are trying to make 10000 tls connection with Kamailio server for SIP signaling.When I try beyond 6500 TLS connection then TLS handshake is failing.We are running the kamailio by using below command - kamailio -f kamailio.cfg -N 40 -m 1024 -M 16 NOTE: In kamailio.cfg, we have mentioned tls_max_connections=60000 Could you please guide me, how can I improve the tls connections with kamailio ? Thanks Priyaranjan

2 1

Information
by Reaz Baksh 02 Sep '15

02 Sep '15

Hello I'm looking for information on setting SER up on Ubuntu. I have it installed along with rtpproxy but cannot get any info from this site as all documentation links are not working. What I want to do is to receive a input rtp live stream and restream it out to various TV's and decoders. Is this possible with SER ? Thanks Reaz

2 1

Re: [SR-Users] unsupported route_type 64
by Admin 02 Sep '15

02 Sep '15

Here is the output from "event_route[evapi:message-received]" Aug 30 13:18:49 everest /usr/local/kamailio/sbin/kamailio[3897]: INFO: <script>: Message received in evapi: {"Event":"CGR_AUTH_REPLY","TransactionIndex":39453,"TransactionLabel":1341515504,"MaxSessionTime":60,"Suppliers":"","Error":""} Aug 30 13:18:49 everest /usr/local/kamailio/sbin/kamailio[3897]: INFO: <script>: Event received in evapi: "CGR_AUTH_REPLY" Aug 30 13:19:00 everest /usr/local/kamailio/sbin/kamailio[3891]: ERROR: tmx [t_var.c:521]: pv_get_tm_reply_code(): unsupported route_type 64 Thanks.

3 2

OT: some github stuff
by Daniel-Constantin Mierla 01 Sep '15

01 Sep '15

Hello, wondering if people here can shed some light on few github things: 1) is possible to have a multi-line project description (the text at the top of https://github.com/kamailio/kamailio)? Maybe a special format (I tried to add \n or <br />, but they were displayed as is, not interpreted). I would like to add a short note and a link to a page with commit guidelines (http://www.kamailio.org/wiki/devel/github-contributions), so new contributors make proper formatted commits. Displying it at the top will make it easy to spot, as opposity to adding to readme, which is shown at the bottom. 2) is there any specific content/marker to add to README to have it interpreted as markdown (without adding .md extensions)? Something like a special first or last line, ... It will be displayed nicer, but renaming will result in changing lot of other places (makefiles, deb/rpm specs, etc...) and somehow README is the traditional approach for the kind of this file name for Linux/Unix apps. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

3 5

Add PSTN gateway with authentication credentials
by Michael Nielsen 01 Sep '15

01 Sep '15

I'm trying to use a PSTN gateway with Kamailio for outbound - and incoming - calls. Everything seems to be routed correctly, except the PSTN gateway denies the call due to missing authentication. How do I add credentials for my PSTN gateway in a secure way? Best regards, Michael

3 6

blocking call in between two kamailio proxy
by sscc 01 Sep '15

01 Sep '15

i have configered kamailio with two domains sscc1.tk and sscc2.tk. it listen on both domains. each domain has users registered with it. user in domain sscc1 1 can call to user in domain sscc2. i want to allow call in between users of same domai. and block call between user of domain sscc1 to domain sscc2. please help -- View this message in context: http://sip-router.1086192.n5.nabble.com/blocking-call-in-between-two-kamail… Sent from the Users mailing list archive at Nabble.com.

3 6

authorize a client in bulk registration
by Al S 01 Sep '15

01 Sep '15

My PBX registered in bulk to Kamailio successfully.PBX --> Register --> kamailioPBX <-- 401 with nonce value <-- kamailioPBX --> Register with nonce and md5 response values --> kamailioBOX <-- 200ok <-- kamailioHowever, when one of the PBX client sends an invite out, kamailio AUTH module does not authorize this client:client (813-111-1111) --> Invite --> kamailio What would be the right way to authorize a 10-digit client to send a call out when a bulk registration and not 10-digit registration is performed. Your assistance in this matter is greatly appreciatedThanks,Al

3 3

Kamailio TLS configuration
by Alexandru Covalschi 30 Aug '15

30 Aug '15

Hello! I'm having problems with Kamailio configuration with TLS. Or, maybe, that's my misunderstanding about how it should work. So, the issue - inbound TLS works just great, I can call everyone in my domain. I have PositiveSSL certificate, so I have such files: calist.crt AddTrustExternalCARoot.crt + COMODORSAAddTrustCA.crt + COMODORSADomainValidationSecureServerCA.crt divided by \n server.key - key server.crt - cert The configuration of tls.cfg [server:default] method = SSLv23 verify_certificate = no require_certificate = no private_key = /etc/ssl/sectel.io.ssl/sip/server.key certificate = /etc/ssl/sectel.io.ssl/sip/server.crt ca_list = /etc/ssl/sectel.io.ssl/sip/calist.crt #crl = /etc/kamailio/crl.pem (however with or without ca_list nothing changes) [client:default] verify_certificate = yes require_certificate = yes And with that configuration when I'm trying to call to ostel.co (public SIP service supporting TLS) from my server I get such error: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed Putting that in tls.cfg: [client:default] verify_certificate = no require_certificate = no Make everything work. Cross-domain calling is essential and I'm just trying to figure out - what's the problem? Is that my certificate, is that ostel.co certificate or it is just the way it should be? Thanks! -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

3 8

Re: [SR-Users] unsupported route_type 64
by Admin 30 Aug '15

30 Aug '15

Thank you for responding to my post. Here it is. include_file "kamailio-cgrates.cfg" request_route { ... switch ($fU) { case 1001: case 1006: $dlg_var(cgrReqType) = "*prepaid"; break; case 1002: $dlg_var(cgrReqType) = "*postpaid"; break; case 1003: $dlg_var(cgrReqType) = "*pseudoprepaid"; break; default: $dlg_var(cgrReqType) = "*rated"; } $dlg_var(cgrTenant) = "cgrates.org"; $dlg_var(cgrAccount) = $fU; $dlg_var(cgrDestination) = $rU; route(CGRATES_AUTH_REQUEST); # Will be answered in CGRATES_AUTH_REPLY #route(CGRATES_LCR_REQUEST); exit; } # Here will land requests after processing them with CGRateS. # Call RELAY or other routes following this route route[CGRATES_AUTH_REPLY] { if $var(CgrError) != "" { xlog("CGR_AUTH_ERROR: $var(CgrError)"); sl_send_reply("503","CGR_ERROR"); exit; } if $var(CgrMaxSessionTime) != -1 { if $var(CgrMaxSessionTime) == 0 { // Not enough balance, do not allow the call to go through sl_send_reply("403","Insufficient credit"); exit; } else if !dlg_set_timeout("$var(CgrMaxSessionTime)") { sl_send_reply("503","CGR_MAX_SESSION_TIME_ERROR"); exit; } } if $var(CgrSuppliers) != "" { # Enforce the supplier variable to the first one received from CGRateS $dlg_var(cgrSupplier) = $(var(CgrSuppliers){s.select,0,,}); } route(RELAY); } ==================================================================== # kamailio-cgrates.cfg # Kamailio-CGRateS related route blocks # Called on new connection over evapi, should normally be the case of CGRateS engine event_route[evapi:connection-new] { $sht(cgrconn=>cgr) = $evapi(srcaddr) + ":" + $evapi(srcport); # Detect presence of at least one connection } # Called when the connection with CGRateS closes event_route[evapi:connection-closed] { $var(connClosed) = $evapi(srcaddr) + ":" + $evapi(srcport); if $sht(cgrconn=>cgr) == $var(connClosed) { $sht(cgrconn=>cgr) = $null; } } # Message received from CGRateS, dispatch it to own route event_route[evapi:message-received] { json_get_field("$evapi(msg)", "Event", "$var(Event)"); route($(var(Event){s.rm,"})); # String characters are kept by json_get_field, remove them here } # Called by Kamailio on new dialog event_route[dialog:start] { route(CGR_CALL_START); } # Called by Kamailio on dialog end event_route[dialog:end] { route(CGR_CALL_END); } # Called by Kamailio on local disconnect event_route[tm:local-request] { route(CGR_CALL_END); } # Send AUTH_REQUEST to CGRateS route[CGRATES_AUTH_REQUEST] { # Auth INVITEs with CGRateS if $sht(cgrconn=>cgr) == $null { sl_send_reply("503","Charging controller unreachable"); exit; } evapi_async_relay("{\"event\":\"CGR_AUTH_REQUEST\", \"tr_index\":\"$T(id_index)\", \"tr_label\":\"$T(id_label)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_setuptime\":\"$TS\", \"cgr_computelcr\":\"true\"}"); } # Process AUTH_REPLY from CGRateS route[CGR_AUTH_REPLY] { json_get_field("$evapi(msg)", "TransactionIndex", "$var(TransactionIndex)"); json_get_field("$evapi(msg)", "TransactionLabel", "$var(TransactionLabel)"); json_get_field("$evapi(msg)", "MaxSessionTime", "$var(MaxSessionTime)"); json_get_field("$evapi(msg)", "Suppliers", "$var(Suppliers)"); json_get_field("$evapi(msg)", "Error", "$var(Error)"); $var(id_index) = $(var(TransactionIndex){s.int}); $var(id_label) = $(var(TransactionLabel){s.int}); $var(CgrMaxSessionTime) = $(var(MaxSessionTime){s.int}); $var(CgrSuppliers) = $(var(Suppliers){s.rm,"}); $var(CgrError) = $(var(Error){s.rm,"}); t_continue("$var(id_index)", "$var(id_label)", "CGRATES_AUTH_REPLY"); # Unpark the transaction } # Send AUTH_REQUEST to CGRateS route[CGRATES_LCR_REQUEST] { # Auth INVITEs with CGRateS if $sht(cgrconn=>cgr) == $null { sl_send_reply("503","Charging controller unreachable"); exit; } evapi_async_relay("{\"event\":\"CGR_LCR_REQUEST\", \"tr_index\":\"$T(id_index)\", \"tr_label\":\"$T(id_label)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_setuptime\":\"$TS\"}"); } # Process LCR_REPLY from CGRateS route[CGR_LCR_REPLY] { json_get_field("$evapi(msg)", "TransactionIndex", "$var(TransactionIndex)"); json_get_field("$evapi(msg)", "TransactionLabel", "$var(TransactionLabel)"); json_get_field("$evapi(msg)", "MaxSessionTime", "$var(MaxSessionTime)"); json_get_field("$evapi(msg)", "Suppliers", "$var(Suppliers)"); json_get_field("$evapi(msg)", "Error", "$var(Error)"); $var(id_index) = $(var(TransactionIndex){s.int}); $var(id_label) = $(var(TransactionLabel){s.int}); $var(CgrMaxSessionTime) = $(var(MaxSessionTime){s.int}); $var(CgrSuppliers) = $(var(Suppliers){s.rm,"}); $var(CgrError) = $(var(Error){s.rm,"}); t_continue("$var(id_index)", "$var(id_label)", "CGRATES_AUTH_REPLY"); # Unpark the transaction } # CGRateS request for session disconnect route[CGR_SESSION_DISCONNECT] { json_get_field("$evapi(msg)", "HashEntry", "$var(HashEntry)"); json_get_field("$evapi(msg)", "HashId", "$var(HashId)"); json_get_field("$evapi(msg)", "Reason", "$var(Reason)"); jsonrpc_exec('{"jsonrpc":"2.0","id":1, "method":"dlg.end_dlg","params":[$(var(HashEntry){s.rm,"}), $(var(HashId){s.rm,"})]}'); #$jsonrpl($var(reply)); } # Inform CGRateS about CALL_START (start prepaid sessions loops) route[CGR_CALL_START] { if $sht(cgrconn=>cgr) == $null { xlog("Charging controller unreachable"); exit; } evapi_async_relay("{\"event\":\"CGR_CALL_START\", \"callid\":\"$dlg(callid)\", \"from_tag\":\"$dlg(from_tag)\", \"h_entry\":\"$dlg(h_entry)\", \"h_id\":\"$dlg(h_id)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_answertime\":\"$TS\", \"cgr_supplier\":\"$dlg_var(cgrSupplier)\"}"); } # Inform CGRateS about CALL_END (stop debit loops, perform accounting if desired in this way) route[CGR_CALL_END] { if $sht(cgrconn=>cgr) == $null { xlog("Charging controller unreachable"); exit; } $var(callDur) = $TS - $dlg(start_ts); evapi_async_relay("{\"event\":\"CGR_CALL_END\", \"callid\":\"$dlg(callid)\", \"from_tag\":\"$dlg(from_tag)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_answertime\":\"$dlg(start_ts)\", \"cgr_duration\":\"$var(callDur)\", \"cgr_supplier\":\"$dlg_var(cgrSupplier)\", \"cgr_disconnectcause\":\"$T_reply_code\"}"); }

2 1

presence_dialoginfo- first NOTIFY - force_dummy_dialog
by evoip＠ukr.net 29 Aug '15

29 Aug '15

||Hello, we use Kamailio 4.2 with Presence_DIALOGINFO module for BLF notifications (blinking lamps on phone panel). Since version 4.1 Kamailio has a new module parametr " modparam("presence_dialoginfo", "force_dummy_dialog", 1)" - this option generate first NOTIFY to Subscribed phone (without PUBLISH message) and it's lamps became GREEN ( <state>terminated</state>). Without this module parametr the first NOTIFY won't be sent to phone and lamps will stay turned off until first PUBLISH will be handled by presence server. The first NOTIFY sip message with xml body look like this: NOTIFY sip:1005@22.22.22.22:1120 SIP/2.0. To: <sip:1005@11.11.11.11>;tag=867246696. From: <sip:9095@11.11.11.11>;tag=495949c4c25d687c77ca707bd10a0818-5171. CSeq: 2 NOTIFY. Call-ID: 662126467-5060-6(a)BJC.BGI.E.CDD. Route: <sip:10.60.1.3;r2=on;lr=on>, <sip:11.11.11.11;r2=on;lr=on>. Content-Length: 272. User-Agent: kamailio (4.2.3 (x86_64/linux)). Max-Forwards: 70. Event: dialog. Subscription-State: active;expires=600. Content-Type: application/dialog-info+xml. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" ух state="full" entity="sip:9095@10.60.1.3"> <dialog direction="recipient"> <state>terminated</state> </dialog> </dialog-info> Especially pay attention on XML body. But for some SIP phones this xml body with state = terminated is not enough to turn the BLF lamp on. (for example Grandstream phones). It requires one more or two more fields: "dialog id", "call-id". <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="28" state="full" entity="sip:9093@10.60.1.3"> <dialog id="830b4fd4-c75d-1233-ca9f-0050569560f4" call-id="830b4fd4-c75d-1233-ca9f-0050569560f4" direction="recipient"> <state>terminated</state> </dialog> </dialog-info> The question is how to modify this first NOTIFY XML body message to add "dialog id" and "call-id" to </dialog>. For example latest Kamailio Kazoo fixed that already. :) Kind Regards Ewgeny

4 8

unsupported route_type 64
by Admin 29 Aug '15

29 Aug '15

Hi, I am running Kamailio (4.4.0-dev2 (x86_64/linux)) with cgrates real-time billing application. At the end of a basic call (prepaid to postpaid), I receive the following error: ERROR:tmx [t_var.c:521]:pv_get_tm_reply_code():unsupported route_type 64 8(6740) DEBUG:tmx [t_var.c:526]:pv_get_tm_reply_code():reply code is <0> What does route_type 64 mean? I checked the archive posts (FS#456) and seemed to be fixed in branch 4.1 (part of 4.1.5). Thanks.

2 1

Parser error when enabling siptrace functionality
by Asgaroth 28 Aug '15

28 Aug '15

Hi All, I have a strange issue occuring, when I enable the siptrace functionality within the routing logic, I begin to get a bunch of parser error as show below. Is this normal behaviour or, more than likely, am I doing something wrong? I enable siptrace as follows: request_route { route("InitialMessageChecks"); sip_trace(); setflag(22); ... } event_route[tm:local-request] { sip_trace(); ... } When I enable this functions in the routing logic, I begin to see alot of the following messages in the log file: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 39) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021® #023ÄÔ#002 ÊÂÕ#035mùLàU{#002> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 39) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021® #023ÄÔ#002 ÊÂÕ#035mùLàUÞž#002> ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 35) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021® #023ÄÔ#002 ÊÂÕ#035mùLàU #011#003> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#006y|#023ÄN˜#036ÂÕ#035múLàUÅ±#012> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) ERROR: <core> [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023Ä#023Ä#012#007> Is there any way I can disable seeing these messages? What are these messages? Any thoughts are appreciated. Thanks

2 4

kamailio_fifo
by Sven Juhkam 28 Aug '15

28 Aug '15

After updating kamailio I get: ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. Any solution? Best regards Sven

3 2

BLF state on register
by Loic Chabert 28 Aug '15

28 Aug '15

Hello everyone, I'm trying to reproduce BLF's asterisk fonctionnality with kamailio. With asterisk, on register, NOTIFY message has been sent to all suscribed users. So LED became green. How can i reproduce this function ? I have try to execute a "pua_set_publish()" on register route, but it's not work. My led stay "black", and no notify message has been sent. I have loaded presence, presence_xml, presence_mwi, presence_dialoginfo, pua, pua_dialoginfo, pua_usrloc. Any other module needed ? Thanks, Regards.

2 10

webrtc not working in different networks
by Achintha 28 Aug '15

28 Aug '15

Hi, I have configured kamailio 4.3.1 with mysql ,rtpproxy,websocket modules , then I tried to call through the two extensions using x-lite phone and get the capture. Call is connected. In that test rtp was going through the rtpproxy server, Then I configured jssip to test webrtc in this setup, call is connected I can communicate through the extensions both video and voice only in LAN. IN different networks call is connected but, voice is not hear (rtp not working). Kindly help me to configure this web rtc with kamailio. -- Best Regards.. Achintha

1 0

dmq_usrloc
by Kelvin Chua 28 Aug '15

28 Aug '15

is there any function to trigger replication? i enabled the module and dmq is working for my htables but usrloc is not replicating. "kamctl ul show" does not show anything i used ngrep to trace the KDMQ packets, i can see the frequent exchanges but nothing related to usrloc. am i doing something wrong? Kelvin Chua

3 13

multiple proxies
by Bruce Lefko 27 Aug '15

27 Aug '15

If I want to have multiple kamailio proxies in front of multiple media servers, can I balance them using plain old DNS behind a domain name? If I wanted a specific proxy to be notified about an entire SIP dialog I could set the record route to use the public IP of the proxy that received the initial INVITE. Could I leverage something like route53 tcp healthchecks to know if I should be failing over to another proxy?

3 2

multiple proxies
by Bruce Lefko 27 Aug '15

27 Aug '15

If I want to have multiple kamailio proxies in front of multiple media servers, can I balance them using plain old DNS behind a domain name? If I wanted a specific proxy to be notified about an entire SIP dialog I could set the record route to use the public IP of the proxy that received the initial INVITE. Could I leverage something like route53 tcp healthchecks to know if I should be failing over to another proxy?

1 0

kamailio presence server using db_mongodb as database
by José Seabra 27 Aug '15

27 Aug '15

Hello all, I would like to test my presence server using db_mongodb as database server instead of using mysql, but i don't find any documentation explaining how i can configure mongodb database(data structure) for presence or even for the other kamailio modules. Can someone guide me in this setup? Best Regards -- Cumprimentos José Seabra

2 12

Confusion about various TCP expiry and lifetime parameters
by M S 27 Aug '15

27 Aug '15

Hi, We have a bit of confusion regarding tcp_connecton_timeout core parameter. The documentation says, -- Lifetime in seconds for TCP sessions. TCP sessions which are inactive for longer than *tcp_connection_lifetime* will be closed by Kamailio. -- However we observe a strange behaviour. 1. The connection is NOT closed by Kamailio unless we additionally set "close_expired_tcp" parameter in usrloc module, 2. Secondly, if we set this parameter to a smaller value say 70 seconds while sip register expiry is 600 seconds (and close_expired_tcp parameter enabled in usrloc module), the connection still remains active (tested after keeping it idle for 120 seconds then sending a sip request on this connection, we expected the request to fail but it does not fails). 3. if we set this parameter to greater then sip register value, e.g. 610 seconds and set close_expired_tcp parameter in usrloc, then we observe disconnect after about 8-10 minutes. Whereas we expect it to continue since user is re-registering every 600 seconds. Can you guys explain why this is happening? What keeps a tcp connection active or makes it inactive? We are using websockets (which use TCP at lower layer) and we observe there is no websocket frame sent or received over the tcp connection, yet it remain active after tcp_connection_lifetime in case 2 above and becomes inactive in case 3. We are using Kamailio v4.3.1 SVN Rev. 4717b5 on Debian Wheezy 32bit OS. Please suggest. Thank you.

2 3

IMS Kamailio "477 Unfortunately error on sending to next hop occurred" code sent by the PCSCF
by Victor Huertas Garcia 26 Aug '15

26 Aug '15

Hello everybody, I have tried to perform a call between alice and bob (both registered in the same SCSCF and from the same PCSCF). The INVITE goes all the way as it is supposed: Alice-->PCSCF-->SBC-->SCSCF-->ICSCF-->SCSCF-->SBC-->PCSCF. But when it reaches PCSCF on Network to User direction, this SIP proxy answers with a "477 Unfortunately error on sending to next hop occurred" code just after an exchange with the RTP Proxy (Offer-Answer). At kamailio.cfg (which it is not change from the default config) this exchange is done under Route[RTPRPROXY] in the function: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove"); Does anybody know why is complaining (see below) about the server on localhost while it is not configured so? Here you can find the logs from PCSCF: Aug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Received command 'offer' from 192.168.0.108:57522Aug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Creating new callAug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Replying to 'offer' from 192.168.0.108:57522Aug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: *** cfgtrace:request_route=[RTPPROXY] c=[/etc/kamailio/pcscf/kamailio.cfg] l=813 a=2 n=returnAug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: *** cfgtrace:request_route=[Term_Initial] c=[/etc/kamailio/pcscf/kamailio.cfg] l=1196 a=24 n=t_relayAug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f480e27ed70,1358,0,192.168.0.105:49284,16): Invalid argument(22)Aug 23 15:10:06 ims /usr/sbin/kamailio[3313]: CRITICAL: <core> [udp_server.c:555]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Here you can find the configuration from RTP proxy (/etc/default/ngcp-rtpengine-daemon) and you can see that I am not listening the localhost port at "INTERFACES": RUN_RTPENGINE=yes LISTEN_TCP=25060LISTEN_UDP=12222LISTEN_NG=22222LISTEN_CLI=9900INTERFACES="192.168.0.108"TIMEOUT=60SILENT_TIMEOUT=3600PIDFILE=/var/run/ngcp-rtpengine-daemon.pidFORK=yes# TOS=184TABLE=0# NO_FALLBACK=yes# PORT_MIN=30000# PORT_MAX=50000# REDIS=127.0.0.1:6379# REDIS_DB=1# B2B_URL=http://127.0.0.1:8090/# LOG_LEVEL=6# LOG_FACILITY=daemon# LOG_FACILITY_CDR=daemon# LOG_FACILITY_RTCP=daemon# NUM_THREADS=5# DELETE_DELAY=30# GRAPHITE=9006# GRAPHITE_INTERVAL=60# GRAPHITE_PREFIX=myownprefix# MAX_SESSIONS=5000 And finally the pcscf.cfg: # SIP / UDPlisten=udp:eth0:4060#listen=udp:127.0.0.1:4070listen=udp:127.0.0.1:4060# SIP / TCP (Monitoring)#listen=tcp:127.0.0.1:4060# SIP / TCP/TLS#listen=tls:109.239.57.200:5061# SIP / Websocket #!define MY_WS_PORT 80 listen=tcp:eth0:MY_WS_PORT # SIP / Websocket/TLS#!define MY_WSS_PORT 443#listen=tls:109.239.57.200:MY_WSS_PORT #alias=pcscf-1.imscore.orgalias=pcscf.home-domain.net#alias=proxy.imscore.org#alias=tls:"wss-proxy.imscore.org":443 # Port, where we listen to Traffic#!define PORT 4060 # NUEVO Workshop!#!define SBCPORT 5080#!define PCSCF_URL "sip:pcscf.home-domain.net"#!define TCP_PROCESSES 32 #!subst "/NETWORKNAME/home-domain.net/"#!subst "/HOSTNAME/pcscf.home-domain.net/"#!define HOSTNAME_IP pcscf.home-domain.net#!define HOSTNAME_ESC "pcscf\.home-domain\.net" # SIP-Address of capturing node, if not set, capturing is disabled.##!define CAPTURE_NODE "sip:10.1.8.55" # Allowed IPs for XML-RPC-Queries#!define XMLRPC_WHITELIST_1 "127.0.0.1"##!define XMLRPC_WHITELIST_2 "127.0.0.1"##!define XMLRPC_WHITELIST_3 "127.0.0.1" # Databases:#!define DB_URL "mysql://pcscf:heslo@127.0.0.1/pcscf"##!define DB_URL2 "con2=>mysql://pcscf:heslo@127.0.0.1/pcscf"##!define DB_URL "con1=>mysql://pcscf:heslo@192.168.5.1/pcscf"##!define DB_URL2 "con2=>mysql://pcscf:heslo@10.1.27.217/pcscf" #! Optional: Server-URL for Websocket-Requests##!define WEBSOCKET_WEBSERVER "phone.imscore.org" # NUEVO Workshop!#!subst "/ICID_VALUE_PREFIX/P-CSCFabcd/"#!subst "/ICID_GEN_ADDR/127.0.0.1/" # IP-Adress(es) of the RTP-Proxy##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222 udp:10.1.27.217:22222"#!define RTPPROXY_ADDRESS "udp:192.168.0.108:22222"## Several features can be enabled using '#!define WITH_FEATURE' directives:## *** To run in debug mode:# - define WITH_DEBUG## *** To enable nat traversal execute:# - define WITH_NAT# - define the connection to the RTP-Proxy: RTPPROXY_ADDRESS## *** To force alls calls through the RTP-Proxy# - this will automagically enable NAT-Traversal# - define FORCE_RTPRELAY## *** To enable IPv4/IPv6 Translation (RTPProxy)# - this will automagically enable NAT-Traversal# - define WITH_RTPIPV4## *** To enable TCP support execute:# - define WITH_TCP## *** To enable TLS support execute:# - adjust CFGDIR/tls.cfg as needed# - define WITH_TLS# - this will automagically enable TCP## *** To enable XMLRPC support execute:# - define WITH_XMLRPC# - this will automagically enable TCP## *** To enable anti-flood detection execute:# - adjust pike and htable=>ipban settings as needed (default is# block if more than 16 requests in 2 seconds and ban for 300 seconds)# - define WITH_ANTIFLOOD## *** To enable the Rx-Interface:# - Configure Rx-Diameter-Interface in pcscf.xml# - define WITH_RX## *** To enable a Homer SIP-Capter-Node:# - define CAPTURE_NODE with a proper address## *** To enable support for the SEMS-SBC# - define WITH_SBC# - configure dispatcher-list with a set of SBC's## *** To enable support for Websocket# - define WITH_WEBSOCKET# - this will automagically enable TCP## Enabled Features for this host:#!define WITH_DEBUG#!define WITH_NAT#!define WITH_NATPING#!define FORCE_RTPRELAY##!define WITH_TLS#!define WITH_XMLRPC#!define WITH_ANTIFLOOD##!define WITH_RX##!define WITH_RX_REG##!define WITH_RX_CALL##!define WITH_TCP##!define WITH_SBC##!define WITH_RTPIPV4#!define WITH_SBC_CALL#!define WITH_REGINFO#!define WITH_WEBSOCKET#!define WITH_IMS_HDR_CACHE Thanks a lot for your attention

2 1

Kamailio dead after INVITE from SIPDROID
by Kowalski Piotr 3 - Hurt 26 Aug '15

26 Aug '15

Hello Community. We have openIMS nodes in our lab. We have Kamailio 4.0 based solution. I registered SIPDROID client (iptel) on Android and JITSI on WINDOWS. When I call from SIPDROID to JITSI following happens: SIP DROID - iptel - client - registration ok, but INVITE attempt towards JITSI (and other clients) does what's below: HSS (every 30 seconds): [root@hss1 kowalp00]# 2015-08-25 12:16:55,468 WARN de.fhg.fokus.diameter.DiameterPeer.transport.Communicator - run Read 3021 before Exception. 2015-08-25 12:16:56,980 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf1.hss.ims.test.pl:3868 not responding to connection attempt 2015-08-25 12:17:26,985 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf1.hss.ims.test.pl:3868 not responding to connection attempt SCSCF: Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8879]: INFO: <core> [main.c:854]: INFO: signal 15 received Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8865]: INFO: <core> [main.c:854]: INFO: signal 15 received Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: INFO: snmpstats [snmpstats.c:410]: The SNMPStats module got the kill signal Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: INFO: snmpstats [snmpstats.c:414]: Shutting down the AgentX Sub-Agent! Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Aug 25 12:25:18 scscf1 /usr/sbin/kamailio[8803]: : <core> [main.c:697]: BUG: shutdown timeout triggered, dying... Kamailio becomes dead and needs restart. Clients become unregistered. A call from JITSI to SIPDROID it's fine. Any ideas? Piotr Kowalski Starszy Architekt Techniczny Tel. +48 22 699 50 93 Kom. +48 519 123 289 Orange Orange Polska Hurt Badania i Rozwój Sieci i Platform Usługowych Wydział Usług Konwergentnych Obrzeżna 7 /p. 309, 02-691 Warszawa Fax. +48 22 857 99 86 <http://www.hurt-orange.pl/> www.hurt-orange.pl - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Czy musisz drukować tę wiadomość? Pomyśl o środowisku. Korzystaj z e-faktury. Treść tej wiadomości jest własnością Orange Polska i zawiera informacje stanowiące tajemnicę przedsiębiorstwa Orange Polska. Jeżeli nie jesteście Państwo jej adresatem, bądź otrzymaliście ją przez pomyłkę, prosimy o powiadomienie o tym nadawcy oraz trwałe jej usunięcie. Orange Polska Spółka Akcyjna z siedzibą i adresem w Warszawie (02-326) przy Al. Jerozolimskich 160, wpisana do Rejestru Przedsiębiorców prowadzonego przez Sąd Rejonowy dla m.st. Warszawy XII Wydział Gospodarczy Krajowego Rejestru Sądowego pod numerem 0000010681; REGON 012100784, NIP 526-02-50-995; z pokrytym w całości kapitałem zakładowym wynoszącym 3.937.072.437 złotych.

2 1

Rewrite Invite w/o sdp
by Serge Yuriev 25 Aug '15

25 Aug '15

Hello, Our PBX (Cisco CallManager) uses "late offer" sending invites w/o sdp. After receiving provisional reply 18x it answers prack with sdp I'm trying to realize following scenario: 1 Pbx (invite no sdp) 2 kamailio hold and saves invite 3 send 183 to Pbx 4 get prack with sdp 5 construct new invite from saved one and sdp from prack 6 send it to calee Is this possible? How to save invite for future use? -- Serge via mobile

3 2

dispatcher gateway probing in 4.3
by Joseph Dickson 24 Aug '15

24 Aug '15

Greetings.. I'm fairly new to Kamailio, and am having fun so far... I'm using the latest 4.3 rpm version on Centos 7. I'm using Kamailio to front end a pair of FreeSwitch SBC boxes in an active/active config.. My plan is to use dispatcher to load balance calls between them. This is working fine.. the problem that I'm bumping into is that the gateway probing options in dispatcher seem not to fit my use case.. In addition to the two SBC gateways, I also have a few external gateways that I want to load balance between also using dispatcher. The problem is that I can't send options packets to these gateways -- they don't respond to them. This means I can't just configure dispatcher to probe all gateways (probing_method=1) because it will mark these gateways as inactive.. It looks like every other configuration of probing_method only probes a gateway until the state is determined, and then it stops probing. probing_method=2 does not appear to be used at all in the code.. Am I doing this wrong? Or is this just an area of dispatcher that needs some improvement? Looking at the code, I can probably add the functionality that I need, but I don't want to go that route if I'm just missing how this is usually done.. Thanks, -- *Joseph Dickson* E: jdickson(a)evolvetsi.com

3 5

Request timout - Change value
by Loic Chabert 24 Aug '15

24 Aug '15

Hello everybody, I'm using kamailio as a dispatcher in front of asterisk boxes and i use a failure route if asterisk box does not respond or send 500error. failure_route[RTF_DISPATCH]{ if(t_is_canceled()){ exit(); } if(t_check_status("500") || (t_branch_timeout() && !t_branch_replied())){ xlog("L_WARN","[$fU@$si:$sp]{$rm} Asterisk Box $du is down\n"); ds_mark_dst("ip"); if(ds_next_dst()){ xlog("L_WARN","[$fU@$si:$sp]{$rm} Sending to Asterisk Box - $du\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } As you can see, i check if "t_branch_timeout() && !t_branch_replied()", but where has been stored timeout value ? How can i change it ? For exemple, i need to set 1s, and then, request expired. Thanks, Regards.

2 4

Dialog module - db_fetch_rows
by jay binks 24 Aug '15

24 Aug '15

Hey All, Im just after someone else thoughts on if this is a bug. After reading : http://kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#idp148408 Im lead to understand that setting *modparam("dialog", "db_fetch_rows", 0)* should instruct the dialog module to write dialogs to the DB, but not bother loading them on startup. This sounds like what I'm after. However ever time I set the value to 0, dialogs are still loaded from the DB. a quick dig in the source takes me to the function select_entire_dialog_table in dlg_db_handler.c this chunk of code does not appear to properly implement " if fetch_num_rows == 0 then do nothing " Im after a couple of things from you guys. a) validation that my understanding of what SHOULD happen is correct b) possibly validating that I'm not crazy, and this is actually a bug :) -- Sincerely Jay

2 4

Ims Kamailio issue - "403 Forbidden - You must register first with a S-CSCF"
by Victor Huertas Garcia 23 Aug '15

23 Aug '15

Hello everybody, I have installed an IMS core using Kamailio and FHoSS. In this environment, I am able to successfully register Bob and Alice with the same S-CSCF. However, when I try to make a voice call between Bob and Alice it is not established. The code received is "403 Forbidden - You must register first with a S-CSCF". This code is generated by the P-CSCF because the Contact Header that is checked against the database ("location" table) it is not found. When I look up manually the "location" MySQL table, there are two things that surprise me: - The Contact header (AOR column content) that is included in the INVITE Requet URI when entering in the P-CSCF is present in the table. So why the pcscf_is_registered ("location") returns false? - When Bob registers two Contacts are stored, the first one is the one included in the REGISTER message generated from Bob IMS Client. However, the second Contact is the one present in the 200 OK message (coming from the S-CSCF) which is different from the one present in the REGISTER (the only thing different is the UDP port). Why the S-CSCF (with the default kamailio.cfg) doesn't the Contact header and changes the UDP port? Is this the standard behaviour? I don't think so... Thanks for your attention Regards,

2 3

A question to you, SIP providers: what is your policy on PBXes calling themselves
by Antonio Gómez Soto 23 Aug '15

23 Aug '15

Hello, This is question on PBX behavior, what is the right thing to do, and how do PBX's generally behave. If a user on a phone, dials a number, which happens to be configured on the same phone system (for example another tenant), there are two options: 1. The PBX notices this, and directly connects the phone to the DID on that system (breaking separation of tenants) 2. The PBX sends the call out on the SIP trunk, and the provider-routing sends the call back as an incoming call. What are the pros and cons of each option from the SIP provider point of view? How do PBX's generally behave? Thanks, Antonio PS: I reposted, because the original question was apparently not phrased in a clear way.

3 2

IPSec supporting open source SIP Server
by Priyaranjan Nayak 23 Aug '15

23 Aug '15

Hi All, I need one IPSec supported opensource SIP server for testing my sip client endpoint.Could you please tell me which SIP server supporting IPSec ? Thanks in advance. Thanks Priyaranjan

5 4

Re: [SR-Users] Presence and NAT
by Pramod Venugopal 21 Aug '15

21 Aug '15

I realize this last email has no useful information, so let me elaborate. I have a Kamailio server on AWS with an internal IP of 10.20.30.40 and it has an external elastic IP of 55.66.77.88. When a SIP client subscribes it sends the initial subscription to user@realm (where realm resolves to 55.66.77.88) , then it gets a notify with a new address <sip:10.20.30.40:5060;transport=udp> . The client attempts to refresh the subscription, but we are now sending the subscribes to this new contact which will time out. 300 seconds later, we subscribe again to user@realm and contacts show up. Other stuff like rtp, inbound/outbound calling all work properly. Pramod Venugopal pramod(a)dvnull.org From: sr-users on behalf of Pramod Venugopal Reply-To: "Kamailio (SER) - Users Mailing List" Date: Thursday, August 20, 2015 at 10:38 PM To: "Kamailio (SER) - Users Mailing List" Subject: [SR-Users] Presence and NAT Hello everyone, I have a Kamailio 4.2 system on Amazon AWS. When subscribing to presence the contact is sent out with the internal IP rather than the external public IP. The subscribes are being sent to it with the external hostname ( which resolves to the external IP ). Where would I be able to change the IP sent in the Contact header ? Pramod Venugopal pramod(a)perseus-tech.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Problems with RTPPROXY module
by Michael Nielsen 21 Aug '15

21 Aug '15

Hello everyone, I'm new to Kamailio and are trying this tutorial: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I've installed everything and are trying to launch Kamailio, but it fails with the following error: ----- systemctl status kamailio.service ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2015-08-21 06:26:26 CDT; 8s ago Process: 1474 ExecStart=/etc/init.d/kamailio start (code=exited, status=1/FAILURE) Aug 21 06:26:26 sip kamailio[1474]: error in config file /etc/kamailio/kamailio.cfg, line 234, column 12-25: failed to load module 0(1483) ERROR: <core> [sr_module.c:563]: load_module(): could not find module <rtpproxy> in </usr/lib64...3432]: yyerror_at(): Aug 21 06:26:26 sip kamailio[1474]: r in config file /etc/kamailio/kamailio.cfg, line 270, column 35: Can't set module parameter 0(1483) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <registrar> found 0(148...ailio/kamailio.cfg, Aug 21 06:26:26 sip kamailio[1474]: : <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(1483) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 295, column ...mod_param_regex(): N Aug 21 06:26:26 sip kamailio[1474]: 1483) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 321, column 36: Can't set module parameter 0(1483) ERROR: <core> [modparam.c:150]: set_mod_para...]: yyerror_at(): par Aug 21 06:26:26 sip kamailio[1474]: file /etc/kamailio/kamailio.cfg, line 378, column 50: Can't set module parameter 0(1483) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command is_method (params 1) 0(1483) : <co...mailio.cfg, line 417 Aug 21 06:26:26 sip kamailio[1474]: fg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 436, column 24: unknown command, missing loadmodule? 0(1483) ERROR: <core> [pvapi.c:810]: pv_parse_spec2(): erro...): wrong char [U/85] Aug 21 06:26:26 sip kamailio[1474]: failed! Aug 21 06:26:26 sip systemd[1]: kamailio.service: control process exited, code=exited status=1 Aug 21 06:26:26 sip systemd[1]: Failed to start LSB: Start the Kamailio SIP proxy server. Aug 21 06:26:26 sip systemd[1]: Unit kamailio.service entered failed state. Hint: Some lines were ellipsized, use -l to show in full. ----- It seems that if it cannot find the module RTPPROXY. I have installed it via apt-get install rtpproxy and it seems to be running. What to do?

3 5

Debian nightly builds
by Jon Bonilla (Manwe) 21 Aug '15

21 Aug '15

Hi all Trying to install latest nightly build of several branches I find that they are from June. Is the repo outdated? http://deb.kamailio.org/kamailiodev-nightly/pool/main/k/kamailio/ cheers, Jon

2 1

Presence and NAT
by Pramod Venugopal 21 Aug '15

21 Aug '15

Hello everyone, I have a Kamailio 4.2 system on Amazon AWS. When subscribing to presence the contact is sent out with the internal IP rather than the external public IP. The subscribes are being sent to it with the external hostname ( which resolves to the external IP ). Where would I be able to change the IP sent in the Contact header ? Pramod Venugopal pramod(a)perseus-tech.com

1 0

Back to the dead communications: No outgoing audio
by Manuel Camargo Lominchar 20 Aug '15

20 Aug '15

After 6 months of resolving this situation: http://lists.sip-router.org/pipermail/sr-users/2015-February/086980.html Without touching any configs, my server is back to a failure situation I'm not sure if I'm facing the exact same thing, but for some strange reason, without introducing any modification in the servers, one leg is not hearing voice (outgoing traffic), while incoming rtp traffic goes fine, so I have to assume another time that some Update has destroyed it I've mostly done the same checks as before. I started to assume that rtp-engine could be failing by the same cause, but I have updated to the latest version and still the same issue This has happened from 2-3 day to now. Any ideas on how to start trying to find the cause? Versions: Kamailio 4.1.4 RTP Engine: 4.1.0.0 Kind regards, *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com

1 1

Mitel 5000 and 3000
by Fred Posner 20 Aug '15

20 Aug '15

I rarely integrate these with Kamailio and am having some "resistance" in assisting. I was wondering if anyone would be willing to share any configuration recommendations for Mitel when using Kamailio as a SIP trunk (outbound/inbound). Sincerely, Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

3 8

Re: [SR-Users] Mitel 5000 and 3000
by rahul.ultimate 20 Aug '15

20 Aug '15

Hi Fred I too am new to Mitel products but the attached pic might be of some use to you. Thanks RM Sent from Samsung Mobile -------- Original message -------- From: Fred Posner <fred(a)palner.com> Date:18/08/2015 7:08 PM (GMT+05:30) To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Mitel 5000 and 3000 On 08/18/2015 09:23 AM, Alex Balashov wrote: > Are there any specific issues you're encountering? > Mostly hold issues where they try to speak directly with the media server vs the sip server; as well as general audio issues in/out. Most of these seem to be acl related with a set-up issue on the Mitel. I don't work with Mitel enough to discuss in "Mitel speak" what to change. Apparently RTP ports is a different language to the people I am working with. =) --fred _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

new SIP account can not be created ... DB Error : connect failed
by Rainer Böing 20 Aug '15

20 Aug '15

Hello, want to build the following account about Ipel.Org . My old following account no longer works. Iptel.org 19.11.2013 E-Mail rainer.boeing(a)t-online.de Username: boe.rai(a)iptel.org <mailto:boe.rai@iptel.org> Password: xd7ghg Sip address: <sip:boe.rai@iptel.org> sip:boe.rai@iptel.org I can also create no new no account . Please Help !!! clamp THANKS Greetings from Germany Rainer Böing Disclaimer: Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

2 1

Change ports for outgoing RTP packets in RTPproxy
by Jean-Marie Baran 19 Aug '15

19 Aug '15

Hi, Is there a way to change the ports were RTPproxy sends outgoing packets ? By rewriting the SDP before it gets to RTPproxy, or anything else ? 1 - Kamailio receives the SDP 2 - It somehow transmits the contact information to RTPproxy (ip + ports) 3 - The client start sending packets to RTPproxy which in turn relays them to the contact How to change the ports in step 2 ? Sincerely, -- *Jean-Marie Baran*

2 2

how to use seturi with varibales in kamailio
by SureshTummala 19 Aug '15

19 Aug '15

Hi, I have the following questions 1. How to remove ruri which is already added to the destination set2. How to rewritehost in ruri with variables3. How to change the To header in INVITE which are generated by t_relay based on destination set Regards,Suresh Tummala

2 1

Kamailio with NATted RTPproxy
by Jean-Marie Baran 18 Aug '15

18 Aug '15

Hi, I am trying to have RTPproxy works with Kamailio behind a NAT. As it seems RTPproxy cannot handle NAT natively, I installed RTPproxy 1.2.1 by miconda: https://github.com/miconda/rtpproxy Now I am able to make calls, but neither the video nor the audio are present. Wireshark showed that the RTP packets were sent to wrong IPs (private IPs). And this was because of the SDP which contained the wrong IP (fields o= and c=). Then I tried to play with rtpproxy_manage(), and Kamailio replaced the IP in o= and c= by its advertising IP (i.e. its public IP), so the packets from the client behind the NAT were sent to the public address of Kamailio (and then dropped). My question here is what should be the addresses in the SDP when redirected from Kamailio to the client ? Here is the topology when receiving an ACK with SDP: Client <--(2)-- Kamailio <--(1)-- Router (NAT) <--(1)-- Public Internet <--(1)-- Third party router (21.24.12.24) (1): SDP o= 21.24.12.24, c= 21.24.12.24 (2): SDP o= ??, c= ?? (Should it be Kamailio public IP (actually Router IP), or Kamailio private IP, or third party SIP server public IP ?) Cheer, -- *Jean-Marie Baran*

2 7

rtpengine rewrite sdp for second time
by Virmantas Variakojis 18 Aug '15

18 Aug '15

Hi Carsten, Thanks for your reply! How can I call msg_apply_changes() in right way? If I call msg_apply_changes() for example in route[RELAY] i get error: ERROR: textopsx [textopsx.c:157]: msg_apply_changes_f(): invalid usage - not in request route if I do everything as you suggested in onreply_route (after I get for example SIP 500 message) it doesn't work and my SDP remains unchanged. Thanks! > > *From: *Carsten Bock <carsten(a)ng-voice.com> > *Subject: **Re: [SR-Users] rtpengine rewrite sdp for second time* > *Date: *17 Aug 2015 09:49:12 EEST > *To: *"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > > > *Reply-To: *"Kamailio $SER$ - Users Mailing List" < > sr-users(a)lists.sip-router.org> > > Hi Virmantas, > > you could try to one or a combination of the following: > - explicitly delete the first session by calling rtpengine_manage() in > the failure route > - call msg_apply_changes() from Textops before calling > rtpengine_manage() the second time (so the already made changes are > applied and the message gets reparsed) > > Thanks, > Carsten > > 2015-08-16 20:13 GMT+02:00 Virmantas Variakojis > <virmantas.variakojis(a)gmail.com>: > > Hi, > > I have a situation like this: > I have kamailio with 3 NIC's. I'm sending INVITE to first provider over > NIC1. Before sending INVITE I use rtpengine_manage to rewrite sdp body with > IP that is on NIC1: > rtpengine_manage ("replace-origin replace-session-connection > direction=intvlan direction=prov1 ICE=remove"); > > If first provider fails i want to send INVITE for the same call to second > provider. But if i use like this: > rtpengine_manage ("replace-origin replace-session-connection > direction=intvlan direction=prov2 ICE=remove"); > or (because ip of prov1 is allready set): > rtpengine_manage ("replace-origin replace-session-connection > direction=prov1 > direction=prov2 ICE=remove"); > rtpengine ignores my changes as said in documentation: "The direction must > only be specified in for initial SDP offer; answers or subsequent offers > can > omit this option." > > and if I try to set ip like this: > rtpengine_manage ("replace-origin replace-session-connection > media-address=X.X.X.X ICE=remove"); > SDP body is doubled: it leaves old values and adds the new one. > > What i'm doing wrong or how can I rewrite SDP for second time? Thanks in > advance for answer! > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 5247593-0 > Fax +49 40 5247593-99 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

Routing re-INVITEs after phone port change
by David Cunningham 18 Aug '15

18 Aug '15

Hello, We have a scenario like this: SUA -> Kamailio with registrar module -> Asterisk A call from the SUA is set up with SIP timers, and after 15 minutes Asterisk sends a re-INVITE to Kamailio to forward on to the SUA. That re-INVITE has a RURI with the address and port of the SUA at the time the call started. Now if the SUA re-registers after the call starts and before the re-INVITE, and is on a new address or port number, then the re-INVITE never gets to the phone. Obviously Kamailio should send the re-INVITE to the new address/port, but is not. The re-INVITE is routed using the lookup() function. Can anyone suggest why this is happening? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

2 5

Kamailio/Asterisk Integration - MySQL driver error
by Ben Fitzgerald 17 Aug '15

17 Aug '15

Has anyone seen this error before? " ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Host '192.168.xxx.xxx is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts'" I fixed it using flush hosts but I'm curious why this popped up? Is there a way to limit connections for Kamailio and why am I reaching that limit with *only 2* SIP endpoints? This is not a heavily utilized database by any means. 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

2 9

Siremis docker
by Sylvain Boily 17 Aug '15

17 Aug '15

Hello, I did a docker image for siremis, if some people want to play easily like me with kamailio and siremis ;) Info here : https://github.com/sboily/dockerfile/tree/master/siremis or : https://hub.docker.com/r/quintana/siremis/ Have a good day. Sylvain

3 2

Edge Proxy, how is it supposed to work?
by Daniel Tryba 17 Aug '15

17 Aug '15

The documentation for outbound contains sample configs for edge proxy and registrar configs: http://kamailio.org/docs/modules/4.2.x/modules/outbound.html I tried to implement these but I don't see anything happen concerning the outbound module. What is supposed to happen to make it work? The main problem I have is that calls for the endpoints coming in from the registrar fail to get past the loose_routing check and fail with a 500 in the default case of the switch. the R-URI contains username/externalip/port from the endpoint, there is 1 route header (containing the edgeproxy ip) but no username/flow-token. So per http://kamailio.org/docs/modules/4.2.x/modules/rr.html#loose-route-id loose_routing returns false: "If a Route: header is found, the function returns 1 and behaves as described in section 16.12 of RFC 3261. There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE." How should the Path header look when registering? According to http://kamailio.org/docs/modules/4.2.x/modules/path.html#idp3043008 there should be a flowtoken inserted, this is not the case. Just a plain Path: <sip:ip;lr> is added. So when is outbound required? Can anybody confirm that the examples work?

2 1

db_mongodb compiling errors
by José Seabra 17 Aug '15

17 Aug '15

Hello all, I'm trying to build db_mongodb module but i'm facing some problems, please take a look at logs messages below: make CC (gcc) [M db_mongodb.so] mongodb_connection.o In file included from mongodb_connection.c:24: *mongodb_connection.h:26:20: warning: mongoc.h: No such file or directory* *mongodb_connection.h:27:18: warning: bson.h: No such file or directory* In file included from mongodb_connection.c:24: mongodb_connection.h:37: error: expected specifier-qualifier-list before ‘mongoc_client_t’ mongodb_connection.c: In function ‘db_mongodb_new_connection’: mongodb_connection.c:48: warning: implicit declaration of function ‘mongoc_init’ mongodb_connection.c:49: error: ‘km_mongodb_con_t’ has no member named ‘con’ mongodb_connection.c:49: warning: implicit declaration of function ‘mongoc_client_new’ mongodb_connection.c:50: error: ‘km_mongodb_con_t’ has no member named ‘con’ mongodb_connection.c: In function ‘db_mongodb_free_connection’: mongodb_connection.c:78: error: ‘km_mongodb_con_t’ has no member named ‘con’ mongodb_connection.c:79: warning: implicit declaration of function ‘mongoc_client_destroy’ mongodb_connection.c:79: error: ‘km_mongodb_con_t’ has no member named ‘con’ make: *** [mongodb_connection.o] Error 1 [root@ccp1 db_mongodb]# locate bson.h /usr/local/include/libbson-1.0/bson.h /usr/local/src/mongo-c-driver/src/libbson/src/bson/bson.h /usr/local/src/mongo-c-driver-1.1.10/src/libbson/doc/html/streaming-bson.html /usr/local/src/mongo-c-driver-1.1.10/src/libbson/src/bson/bson.h [root@ccp1 db_mongodb]# locate mongoc.h /usr/local/include/libmongoc-1.0/mongoc.h /usr/local/src/mongo-c-driver/src/mongoc/mongoc.h /usr/local/src/mongo-c-driver/tests/test-libmongoc.h /usr/local/src/mongo-c-driver-1.1.10/src/mongoc/mongoc.h /usr/local/src/mongo-c-driver-1.1.10/tests/test-libmongoc.h I have installed mongo-c-driver successfully and the header files(bson.h, mongoc.h) are located at /usr/local/src/mongo-c-driver/src/mongoc. Can someone help me? -- Cumprimentos José Seabra [image: ITCENTER Store] <http://store.itcenter.com.pt/>[image: ITCENTER Helpdesk] <http://support.itcenter.com.pt/otrs/customer.pl>[image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt>[image: ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter>[image: ITCENTER Twitter] <https://twitter.com/ITCenter_PT>

2 3

rtpengine rewrite sdp for second time
by Virmantas Variakojis 17 Aug '15

17 Aug '15

Hi, I have a situation like this: I have kamailio with 3 NIC's. I'm sending INVITE to first provider over NIC1. Before sending INVITE I use rtpengine_manage to rewrite sdp body with IP that is on NIC1: rtpengine_manage ("replace-origin replace-session-connection direction=intvlan direction=prov1 ICE=remove"); If first provider fails i want to send INVITE for the same call to second provider. But if i use like this: rtpengine_manage ("replace-origin replace-session-connection direction=intvlan direction=prov2 ICE=remove"); or (because ip of prov1 is allready set): rtpengine_manage ("replace-origin replace-session-connection direction=prov1 direction=prov2 ICE=remove"); rtpengine ignores my changes as said in documentation: "The direction must only be specified in for initial SDP offer; answers or subsequent offers can omit this option." and if I try to set ip like this: rtpengine_manage ("replace-origin replace-session-connection media-address=X.X.X.X ICE=remove"); SDP body is doubled: it leaves old values and adds the new one. What i'm doing wrong or how can I rewrite SDP for second time? Thanks in advance for answer!

2 1

multi-process not multi-threading
by Ali Taher 14 Aug '15

14 Aug '15

Hello, Can you please help me to well understand this statement : Kamailio is a multi-process application not multi-threading Does this mean that if we have a machine with one processor of 8 cores , kamailio cannot process more than 1 call at the same time ? Any help would be appreciated. Regards, Ali

3 3

Handling Hold
by Fred Posner 14 Aug '15

14 Aug '15

Hello all, Always fun integrating with proprietary pbx's... is there a way that is "generally accepted" as handling hold invites from devices such as Mitel or Toshiba? We're not proxying the media, so these devices are trying to send an Invite to the endpoint when calls are placed on hold. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

3 9

CNXCC
by Keith 14 Aug '15

14 Aug '15

Hi All, I have CNXCC setup, it is kind of working but when I run kamcmd cnxcc.check_client the call information is still there even when the call is disconnected. Is this right? I basically want to make sure a customer can only have x number of simultaneous calls. Thanks Keith

1 0

Re: [SR-Users] cnxcc
by Keith 13 Aug '15

13 Aug '15

Hi Guys, Trying to use cnxcc but cnxcc.so is not in the modules folder. What is the easiest way to install? Thanks Keith

1 0

Problems investigating a crash in the siptrace module (4.3.1)
by Sven Neuhaus 13 Aug '15

13 Aug '15

Hi, we're experiencing crashes in the siptrace.so module with Kamailio 4.3.1. I looked at the core dump with gdb but it complained about missing debugging symbols so all I could see was that it called do_action() before branching into an unknown function in the siptrace module. I then installed the kamailio-dbg package to get the debugging symbols. This had the weird effect that get the error "Cannot access memory at ..." in GDB and no longer see *any* function names (previously only the names in the siptrace module were missing. Any idea whats going on? # gdb /usr/sbin/kamailio core GNU gdb (GDB) 7.4.1-debian Copyright (C) 2012 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/sbin/kamailio...Reading symbols from /usr/lib/debug/.build-id/e7/8dc57031d294f320b360d8f2f875a005db757e.debug...done. done. [New LWP 31426] Cannot access memory at address 0x3840f881f9c13908 Cannot access memory at address 0x3840f881f9c13900 (gdb) list 1771 main.c: No such file or directory. (gdb) info threads Id Target Id Frame * 1 LWP 31426 0x00007fbc89403f0f in ?? () (gdb) bt full #0 0x00007fbc89403f0f in ?? () No symbol table info available. #1 0x0800011000d46460 in ?? () No symbol table info available. #2 0xffffffff00000001 in ?? () No symbol table info available. #3 0x0000000000d63230 in ?? () No symbol table info available. #4 0x00000000d59d000a in ?? () No symbol table info available. #5 0x0000000000000000 in ?? () No symbol table info available. # dpkg -l kamailio\* Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend |/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad) ||/ Name Version Architecture Description +++-==============================-====================-====================-================================================================= ii kamailio 4.3.1+wheezy amd64 very fast and configurable SIP proxy un kamailio-berkeley-modules <none> (no description available) un kamailio-carrierroute-modules <none> (no description available) un kamailio-cpl-modules <none> (no description available) ii kamailio-dbg:amd64 4.3.1+wheezy amd64 very fast and configurable SIP proxy [debug symbols] un kamailio-ldap-modules <none> (no description available) un kamailio-lua-modules <none> (no description available) un kamailio-mysql-modules <none> (no description available) un kamailio-perl-modules <none> (no description available) ii kamailio-postgres-modules:amd6 4.3.1+wheezy amd64 PostgreSQL database connectivity module for Kamailio un kamailio-presence-modules <none> (no description available) un kamailio-python-modules <none> (no description available) un kamailio-radius-modules <none> (no description available) un kamailio-snmpstats-modules <none> (no description available) un kamailio-tls-modules <none> (no description available) un kamailio-unixodbc-modules <none> (no description available) un kamailio-xml-modules <none> (no description available) un kamailio-xmpp-modules <none> (no description available) Regards, -Sven

2 2

Kamailio SCTP with CentOS
by Mellow Yellow 13 Aug '15

13 Aug '15

Hey guys, I've been pulling out my hair for a few days trying to get SCTP with Kamailio working in CentOS 7... with no luck at all. When attempting to start Kamailio via the 'kamctl start' command I get the following error: INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed When attempting to start Kamailio via the 'systemctl start kamailio' command, I get the following: Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. When checking the 'systemctl status kamailio -l' command, I get the following: kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Thu 2015-08-13 00:42:51 AEST; 14s ago Process: 19229 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: INFO: usrloc [hslot.c:51]: ul_init_locks(): locks array size 1024 Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/lib64/kamailio/modules/ctl.so) Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 13 00:42:51 Elastix5.workgroup kamailio[19229]: Starting kamailio: Aug 13 00:42:51 Elastix5.workgroup systemd[1]: kamailio.service: control process exited, code=exited status=1 Aug 13 00:42:51 Elastix5.workgroup systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Aug 13 00:42:51 Elastix5.workgroup systemd[1]: Unit kamailio.service entered failed state. So I started to think there was an issue with the SCTP config or whether SCTP was enabled in CentOS. However, everything looks fine: # cat /proc/modules sctp 217455 2 - Live 0xffffffffa03ed000 # /sbin/lsmod | grep sctp sctp 217455 2 libcrc32c 12644 2 xfs,sctp # more /proc/sys/net/sctp/addip_enable 1 # checksctp SCTP supported SCTP modules installed properly: Package lksctp-tools-devel-1.0.13-3.el7.x86_64 already installed and latest version Package lksctp-tools-doc-1.0.13-3.el7.x86_64 already installed and latest version Package lksctp-tools-1.0.13-3.el7.x86_64 already installed and latest version Running the ' sctp_test' command between two servers, SCTP is working fine... I can capture SCTP packets between the servers. I've left the Kamailio.cfg file as is... with the only modification being made to enable the SCTP module to start: enable_sctp = 1 Checking 'kamailio -v' USE_SCTP is enabled: # kamailio -v version: kamailio 4.3.1 (x86_64/linux) f38e67 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: f38e67 compiled on 18:17:56 Jul 20 2015 with gcc 4.8.2 I've tried installing Kamailio with the precompiled rpms for CentOS 7, and I also followed some guides on installing in via GIT & compiling it myself... however the issue still remains. Any help would be greatly appreciated as I've completely run out of ideas, and there isn't much info on the web about Kamailio & SCTP. Thanks, Rob

2 2

upgrade V4.2.5 to V3.1 -Build error
by Yasin CANER 13 Aug '15

13 Aug '15

3 3

XCap Server + Postgre
by Igor Olhovskiy 12 Aug '15

12 Aug '15

Hi! I'm experience really strange behavior of Kamailio. Right now it's using postgre database When trying to implement XCAP server capabilities, according to http://kamailio.org/docs/modules/4.3.x/modules/xcap_server.html#idp76784 on a block if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); exit; } While trying to test it via browser (user test, pass test) I've got these errors in log Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: db_postgres [km_dbase.c:247]: db_postgres_submit_query(): 0x7f3fb0660b10 PQsendQuery Error: ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012 Query: select password from subscriber where extension='test' AND user_context='xcap' Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Without this block, I don't get user auth. Is it a bug or it's mine misconfiguration? -- Best regards, Igor

2 3

kamailio as SIP Agent
by Sandeep Chakravarthi 12 Aug '15

12 Aug '15

Hi, We have a requirement with one of our telco We are using asterisk in our servers and we are planning to implement SIP-I protocol and we choosed kamailio for it. In Kamailio website, i came to know that kamailio will be supporting both SIP-I and SIP-T protocols Below is what we need and pls confirm whether it is possible or not? Asterisk PBX <-------> Kamailio <--------> Telco MSC Telco will be forwarding the calls to kamailio on sip-i protocol and kamailio server has to forward the calls to our Asterisk server by converting sip-i to standard sip protocol Similiarly Asterisk will be initiating sip call to kamailio server and kamailio server should convert it into SIP-I and should forward the call to Telco MSC 1. I am able to establish the SIP trunk [sending OPTIONS from asterisk and kamailio acknowledges with 200 OK] between Asterisk and Kamailio using dispatcher module in kamailio and sip.conf in asterisk. How to establish the SIP trunk between kamailio and telco MSC? [Generally MSC will act as SIP server and kamalio should send OPTIONS packet and MSC will acknowledges with 200 OK] My telco MSC has only provided me the MSC SIP IP and there were no username/passwords provided. Means i need to use IP based authentication for the SIP Trunk establishment. In Kamailio how to achieve it? Please help and any suggestions/feedback will be highly appreciated and thankful Regards, Sandeep

2 12

sdp manipulation in connection with rtpengine_offer/answer calls
by Juha Heinanen 12 Aug '15

12 Aug '15

it was not clear to me based on rtpengine module readme if rtpengine_offer/answer are same kind of sdp manipulation functions as, for example, sdpops module functions, i.e., will rtpengine functions pass to rtpengine the sdp as it is after possible earlier sdpops function calls and will possible sdpops functions calls after rtpengine function call keep on manipulating the sdp returned from rtpengine function call. -- juha

4 16

RTPProxy incorrect port 0 in reply from rtp proxy
by Grant Bagdasarian 12 Aug '15

12 Aug '15

Hello, I've been load testing kamailio(4.2) with rtpproxy(2.0) using sipp, and when I hit 50+ CPS I start getting the following error: "incorrect port 0 in reply from rtp proxy". I'm running 4 instances of rtpproxy on the same instance as kamailio. Could somebody tell me why I'm receiving this error? Do I need to run more instances of rtpproxy? # <#----------RTPPROXY Parameters----------#> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7721 udp:127.0.0.1:7722 udp:127.0.0.1:7723 udp:127.0.0.1:7724") In request_route: if (is_method("INVITE")) { record_route(); if (has_body("application/sdp")) { rtpproxy_manage(); } } In onreply_route: if (has_body("application/sdp")) { rtpproxy_manage(); } rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy1.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7723 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy2.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7724 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy3.pid -l 10.0.0.1 Regards, Grant

1 1

Re: [SR-Users] LCR Routing - Cost based routing
by DanB 12 Aug '15

12 Aug '15

Hi Alexandru, Well the concept is simple: on INVITE, you will issue a special format evapi request. You will receive the list of suppliers in the reply (which is received in a separate route) and you continue processing from there. These suppliers are simple tags so your creativity will be the limit of what you can do with them - eg: direct ip addresses received or tags of suppliers which you can further translate with drouting module or any alias related one. For more information on LCR please find this year's presentation on Kamailio World (which I strongly recommend to be attended): https://www.youtube.com/watch?v=Hsvcwleb-fY You can also find the kamailio configuration which I have used in the worshop here: https://github.com/cgrates/cgrates/tree/master/data/tutorials/kamevapi/kama… Let me know if further questions (or join our irc or mailing list so we do not create too much noise here). DanB

3 2

Help with sip balancer
by Bruno 11 Aug '15

11 Aug '15

Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer. Here's my network topology: +---> [asterisk1] [public_ip] | 10.50.10.131 [router] <---NAT---> [kamailio] <---+ 10.50.10.1 10.50.10.120 | +---> [asterisk2] 10.50.10.132 In my setup i planned to use UAC and DISPATCHER modules. I started from the "kamailio-basic.cfg" and added some extra lines to handle UAC and DISPATCHER. All is working fine when i do a test call from a softphone inside network 10.50.10.0/24. When a call is coming from the sip carrier, troubles occurs because asterisk boxes are sending their internal ip in SDP. I understand that i need to rewrite SDP in that case, but i actually don't know how/where. I've attached kamailio configuration and a sip trace taken with sngrep where the problem is visible. For security reasons, i would like to force the RTP through RTPProxy. I'm missing something, and need your help me to understand my errors. Best Regards, Bruno

3 3

Trailing newline with {re.subst,expression}
by Daniel Tryba 11 Aug '15

11 Aug '15

I'm unable to get rid of the \n in "multiline" content of a var. xlog("var: $var(sdp)"); $var(sdp)=$(var(sdp){re.subst,/^a=rtcp:.*//}); xlog("var: $var(sdp)"); a=sendrecv#015#012a=rtcp:31543#015#012a=X-foo:bar becomes: a=sendrecv#015#012#012a=X-foo:bar Using the s flag results in no substitution at all, /^a=rtcp:.*//s should result in a=sendrecv#015#012 according to http://www.kamailio.org/wiki/cookbooks/4.3.x/transformations#resubst_expres… Is this a bug or a feature? If it is a feature, how to get rid of the newline (without additional regexps substitutions).

1 0

force_send_socket/$fs behaviour when binding to INADDR_ANY
by Alex Balashov 11 Aug '15

11 Aug '15

Hello, When binding Kamailio to 0.0.0.0, Kamailio no longer recognises any specific IP address homed on the system as being a "socket" for purposes of forcing traffic out of any specific interface (i.e. setting $fs), or any other purpose for which ingress and egress "sockets" are tracked (e.g. double RR). As I understand it, this is because Kamailio considers only addresses explicitly specified via the 'listen' core config directive to be valid "send socket" arguments. Kamailio _will_ accept addresses already attached to the interface if it is bound interface-wise, e.g. listen=eth1:5060 but not when addresses are added to the interface without restarting the proxy: # ip addr add dev eth1 172.30.110.10/24 Aug 10 17:31:35 centosity6 /usr/local/sbin/kamailio[28627]: WARNING: pv [pv_core.c:2285]: pv_set_force_sock(): no socket found to match [udp:172.30.110.10:5060] Is there any straightforward way to modify this behaviour, so that it would be possible to dynamically add addresses to existing physical interfaces and get Kamailio to utilise them as if they were bound at boot time? Thanks, -- 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/

2 1

Re: [SR-Users] Unable to limit call duration with timeout_avp
by Admin 10 Aug '15

10 Aug '15

Thanks Alex. That worked.

1 0

Unable to limit call duration with timeout_avp
by Admin 10 Aug '15

10 Aug '15

Hi, I am using dialog time_avp() parameter to limit the call duration to a configured number of seconds but unfortunately when the number is reached, i get a warning but the call is not torn down. Here is the output from the log file 3(3711) DEBUG: <script>: return value = 30 -----> $avp(i:10) 5(3713) WARNING: dialog [dlg_handlers.c:1426]: dlg_ontimeout(): timeout for dlg with CallID '1-3743(a)192.168.1.100' and tags '37431' '37401' 5(3713) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): unref dlg 0x7f32ee941140 with 2 -> 0 5(3713) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): ref <=0 for dialog 0x7f32ee941140 5(3713) DEBUG: dialog [dlg_hash.c:320]: destroy_dlg(): destroying dialog 0x7f32ee941140 (ref 0) 5(3713) DEBUG: dialog [dlg_hash.c:336]: destroy_dlg(): removed timer for dlg 0x7f32ee941140 [773:2813] with clid '1-3743(a)192.168.1.100' and tags '37431' '37401' Here is my config. file related to dialog module: #!define DLG_FLAG 4 loadmodule "dialog.so" modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "default_timeout", 3600) # 1 hour timeout modparam("dialog", "db_mode", 0) modparam("dialog", "hash_size", 2048) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "default_timeout", 3600) # 1 hour timeout modparam("dialog", "timeout_avp", "$avp(i:10)") if (is_method("INVITE")) { #$var(rv) = perl_exec("max_allowed_time"); $var(rv) = 30; $avp(i:10) = $var(rv); xlog("L_DBG", "return value = $var(rv)\n"); setflag(FLT_ACC); # do accounting setflag(DLG_FLAG); } Maybe i am missing some configuration related to the dialog module. Any help is very much appreciated. Thanks.

2 2

Handling a non-standard SIP port
by Jean-Marie Baran 10 Aug '15

10 Aug '15

Hi, This is a question for the sake of comprehension, as I actually solved the problem with a workaround. I am trying to get Kamailio working with a third party SIP provider. The SIP server of this provider pings my Kamailio with OPTIONS messages. Being located in a sub-network, the SIP messages are relayed to a router (not a SIP router, just a gateway to the Internet) which is not listening/sending on port 5060 but on 13377. When Kamailio answer to OPTIONS, it send back the message to the router on the port 5060 which is wrong. Here is the topology: *Kamailio* *Router* *SIP Server* /my.sip.domain 174.37.217.1 197.17.144.4/ | | ? <-------------------- 5060 | | 5060 <---------------- 13377 | | | 5060 ----------------−> 5060 | | | | X | You can see that, Kamailio answers to the wrong port, but I think this is somewhat coherent since the SIP message says that the port 5060 is in use. But still, I would have liked that the correct port was used: *Kamailio* *Router* *SIP Server* /my.sip.domain 174.37.217.1 197.17.144.4 / | | ? <-------------------- 5060 | | 5060 <---------------- 13377 | | | 5060 ---------------−> 13377 | | | | ? --------------------> 5060 | But I do not understand where Kamailio is supposed to find the correct port. Below are the transport frames, and SIP frames. Kamailio receives: Internet Protocol Version 4, Src: 174.37.217.1 (174.37.217.1), Dst: 172.16.8.61 (172.16.8.61) Source: 174.37.217.1 (174.37.217.1) Destination: 172.16.8.61 (172.16.8.61) User Datagram Protocol, Src Port: 13377 (13377), Dst Port: sip (5060) Source port: 13377 (13377) Destination port: sip (5060) OPTIONS sip:my.sip.domain:5060 SIP/2.0 Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bKb9340cb29ba From: <sip:197.17.144.4>;tag=1797904074 To: <sip:my.sip.domain> Date: Mon, 10 Aug 2015 10:01:51 GMT Call-ID: c7f38580-5c817680-b94-38512c6(a)197.17.144.4 User-Agent: Cisco-CUCM10.5 CSeq: 101 OPTIONS Contact: <sip:197.17.144.4:5060> Max-Forwards: 0 Content-Length: 0 ...and then sends: Internet Protocol Version 4, Src: 172.16.8.61 (172.16.8.61), Dst: 174.37.217.1 (174.37.217.1) Source: 172.16.8.61 (172.16.8.61) Destination: 174.37.217.1 (174.37.217.1) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) SIP/2.0 200 OK Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bKb9340cb29ba;received=174.37.217.1 From: <sip:197.17.144.4>;tag=1797904074 To: <sip:my.sip.domain>;tag=b27e1a1d33761e85846fc98f5f3a7e58.de7a Call-ID: c7f38580-5c817680-b94-38512c6(a)197.17.144.4 CSeq: 101 OPTIONS Allow: INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,REFER,SUBSCRIBE,NOTIFY Accept: */* Accept-Encoding: Accept-Language: en Supported: Server: kamailio (4.3.1 (x86_64/linux)) Content-Length: 0 We see clearly that the port is wrong here. But in the received SIP frame, everything suggests that the correct port is 5060, and I do not think Kamailio will look at the transport frame to send the response back to the correct port. Am I correct stating this ? Maybe the Via header should contain a port along with the IP address ? I solved the problem by using TCP instead of UDP, as I know that since the TCP connection is maintained, the port would be correct (because imposed by the gateway). I am sorry if the question does not directly relates to Kamailio, but any explanation here would be appreciated, as I am a student trying to learn how all this is working :) Cheers, Jean-Marie.

2 2

rtpengine disabling offers of rtcp
by Daniel Tryba 10 Aug '15

10 Aug '15

Maybe a bit offtopic, but I can't find a rtpengine mailinglist and know the Sipwise guys are reading this list. Since upgrading to 4.2.x and rtpengine instead of rtpproxy I'm seeing problems with calls being terminated by upstream provider after a bit more than 5m. The INVITE process looks okay, the only difference with before as far as I can see is the rtcp attribute in the SDP body. According to http://kamailio.org/docs/modules/4.2.x/modules/rtpengine.html#rtpengine.f.r… and https://github.com/sipwise/rtpengine rtcp offering can be disabled. But neither rtcp-mux-demux nor rtcp-mux-accept have any impact on the adding of the a=rtcp:port attribute in SDP. rtpengine_manage("trust-address replace-origin replace-session-connection rtcp-mux-accept ICE=remove"); or rtpengine_manage("trust-address replace-origin replace-session-connection rtcp-mux-demux ICE=remove"); have no effect op the INVITE to the upstream. kamailio: 4.2.5 rtpengine: 4.1.0.0+0~mr4.1.0.0

3 4

Debian/jessie and 4.3.1 default config doesn't work
by Daniel Tryba 10 Aug '15

10 Aug '15

It looks like the compile defaults of the ctl and mi_fifo module and the default kamailio.conf/kamctlrc conflict. kamailio tries to open: /tmp/buildd/kamailio-4.3.1+jessie/debian/kamailio/var/run/kamailio//kamailio_ctl /tmp/buildd/kamailio-4.3.1+jessie/debian/kamailio/var/run/kamailio/kamailio_fifo and will fail to run, unless explicitly set to conf paths are set to /var/run/kamailio/kamailio_(ctl|fifo)

2 3

Re: [SR-Users] LCR Routing - Cost based routing
by DanB 10 Aug '15

10 Aug '15

Guys, Since I saw the thread growing, I am around if you got questions on CGRateS related LCR, fully compatible with Kamailio via evapi module. DanB

2 1

Running Kamailio 4.3 on Debian 8.1 and systemd
by Grant Bagdasarian 10 Aug '15

10 Aug '15

Hello, I just installed the latest stable release of Kamailio (I believe 4.3.1) on a debian 8.1 system using the install guide (http://www.kamailio.org/wiki/install/4.3.x/git). Starting kamailio didn't work using /etc/init.d/kamailio start. I got the following error: # /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceFailed to start kamailio.service: Unit kamailio.service failed to load: No such file or directory. failed! After some googling I found out I have to add a new file named kamailio.service to the /etc/systemd/system/ directory. Kamailio.service: [Unit] Description=Kamailio SIP Server [Service] Type=forking PIDFile=/var/run/kamailio.pid ExecStart=/usr/local/sbin/kamctl start ExecRestart=/usr/local/sbin/kamctl restart ExecStop=/usr/local/sbin/kamctl stop [Install] WantedBy=multi-user.target Now I can start kamailio using the command: service kamailio start Is this the correct way of doing this? Does the file need any changes? Regards, Grant

1 0

Wrong Call-ID and tags in response to OPTIONS
by Jean-Marie Baran 10 Aug '15

10 Aug '15

Hi, I am trying to get Kamailio working with a third party SIP provider. This provider pings Kamailio with OPTIONS messages, but Kamailio does not answer correctly: the Call-ID and the tags change where they should not. Here is an example: Third party SIP server: OPTIONS sip:false.sip.provider:5060 SIP/2.0 Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bK8e45ac1463a From: <sip:197.17.144.4>;tag=1669836943 To: <sip:false.sip.provider> Date: Fri, 07 Aug 2015 12:34:37 GMT Call-ID: a012e080-5c41a5ce-8e5-38512c6(a)197.17.144.4 User-Agent: Cisco-CUCM10.5 CSeq: 101 OPTIONS Contact: <sip:197.17.144.4:5060> Max-Forwards: 0 Content-Length: 0 Kamailio's answer: SIP/2.0 200 OK Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bK93e5f072f5f;received=174.37.217.1 From: <sip:197.17.144.4>;tag=298911055 To: <sip:false.sip.provider>;tag=b27e1a1d33761e85846fc98f5f3a7e58.8c24 Call-ID: 2defcb00-5c41aa17-93f-38512c6(a)197.17.144.4 CSeq: 101 OPTIONS Server: kamailio (4.3.1 (x86_64/linux)) Content-Length: 0 You can see that the Call-ID change from "a012e080-5c41a5ce-8e5-38512c6" to "2defcb00-5c41aa17-93f-38512c6". Likewise, the From tag change ("1669836943" to "298911055"). What could be the reason for this ? Moreover I think Kamailio should add the field Allow (with INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,REFER,SUBSCRIBE,NOTIFY), but it does not. I should mention that I had to modify /kamailio.cfg/ to have Kamailio answer to OPTIONS messages (I added *the code in bold*): if (*!is_method("OPTIONS") && *!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { *#* sl_send_reply("200","Keepalive"); *options_reply();* exit; } Maybe this hack replaces an expected behavior that I failed to see/configure ? Thanks, Jean-Marie.

3 3

Kamailio 4.3 and XCAP
by Safdar Khan 10 Aug '15

10 Aug '15

Hello sir/mam How can i configure xcap with kamailio 4.3. I am finding difficulties in routing logic of xcap_server module amd xcap_client module. Will you please share proper documentation of these two technology together.

1 0

LCR Routing - Cost based routing
by Arun Kumar 09 Aug '15

09 Aug '15

Hi is there any way to achive following usecase using carrier route / LCR / droude module i have 10 different termination gateways, each gateways will have more than 10000 destinations and each gateways will have different rate, for example 91 Destination ( there will be 10000 entries like 91 ) gw1 gw2 gw3 gw4 gw5 0.001 0.007 0.003 0.002 0.003 now need to logic to load the these destination with cost and sort gateways in order of longest match of dialled number,cost, priority for the dialled destination, which will not use too much CPU and memory Regards Arun

4 9

Kamailio crash when imc_list_rooms
by SamyGo 09 Aug '15

09 Aug '15

Hi, I'm working with IMC module and noticed that when I execute fifo command "imc_list_rooms" it crashes. [root@kamailio75 ]# kamctl fifo imc_list_rooms my Kamailio version is 4.2.1 Right now the way IMC module works is that a SIP UA has to send a chat room command to create a new room, get participants added etc. If I try to add users directly from DB the kamailio don't recognize the room or participants unless restart. Would it be possible to reload the DB tables for IMC via MI command, a big plus if we can do room creation, user addition via MI as well ! Thanks, Sammy

1 0

acc Module missing BYE in database
by Dirk Teurlings - SIGNET B.V. 07 Aug '15

07 Aug '15

Hi again, Running kamailio 4.2.5 with the acc module to do accounting in a MySQL database. We are now running into the issue that sometimes there is no BYE line when there is an INVITE. Digging into this I noticed that the dialog:end event does get triggered, and that CNXCC consumes the call properly. So the BYE message does get sent, but acc doesn't seem to pick it up some of the times. It occurs around maybe once every 100 calls. Anybody got any ideas about this? -- Cheers, Dirk Teurlings

2 3

Kamailio example trace needed UAR/UAA
by Jan Devai 07 Aug '15

07 Aug '15

Hi Experts, we have trouble with the last SIP Registration from I-CSCF to S-CSCF, see below: *The flow is as follows:* UE HSS P-CSCF I-CSCF S-CSCF -----------Sip Register-----------> -------Sip Register-------> <--------------------------UAR--------------- --------------------------UAA---------------> ----SIP Register---> Almost a week ago, I've open a thread about this issue called "Error forwarding to SCSCF", and so far we did not get any further with it :-( I was wondering does somebody have similar Signaling that he could share with us? It could be either with an external or OpenHSS, does not matter at this stage. It would definitely move us forward with at least the signaling comparison as a start point. Maybe the UAA answer and anything happening before is wrong. Any reference, similar working trace would be greatly appreciated. Thank you in advance Jan

2 1

About Acc_radius rad_sttaus Function
by Yasin CANER 07 Aug '15

07 Aug '15

3 4

SDP IPv4 has been concatenate - no audio
by Loic Chabert 07 Aug '15

07 Aug '15

Hello everyone, I'm trying put kamailio in front of asterisk server farm. Fow now, 2 asterisk servers are running and i'm trying to make some basic calls between two UACc. All asterisk servers has been ofuscaded from public internet using 10.189.122.0/24 network. All trafic must be passed throught asterisk so RTPproxy is used to (and used for rtp bridging). Kamailio and rtpproxy is running with public IP address, and private ip address (mhomed=1) But a wired thing append on my SDP body: c line have two rtpproxy public addresses concatenate (see my capture attached). Any reason for this ? Only invite method from my asterisk contains 2 publics IP addresses concatenated. Does it mean than rtp_manage as been executed twice ? Thanks in advance. Regards.

3 6

Problem adding json module
by Joao Alves 07 Aug '15

07 Aug '15

Hi, Have just installed the kamailio 4.3 (from github) and was trying to include the additional modules that I require. I've edited the modules.lst and changed the extra modules to compile "include_modules= db_mysql utils json". During compilation (using "make all") getting the following warnings in relation with the json module. CC (gcc) [M json.so] json_funcs.o json_funcs.c: In function 'json_get_field': json_funcs.c:58:2: warning: implicit declaration of function 'is_error' [-Wimplicit-function-declaration] json_funcs.c:64:2: warning: 'json_object_object_get' is deprecated (declared at /usr/local/include/json-c/json_object.h:303) [-Wdeprecated-declarations] And later, when running the config file: ERROR: <core> [sr_module.c:574]: load_module(): could not open module </usr/local/lib64/kamailio/modules/json.so>: libjson-c.so.3: cannot open shared object file: No such file or directory 0(2344) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 195, column 12-20: failed to load module ERROR: bad config file (1 errors) I have already installed json-c modules and confirmed this installation by doing: root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --cflags json-c -I/usr/local/include/json-c root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --libs json-c -L/usr/local/lib -ljson-c root@ID14337:/usr/local/include/json-c# ls arraylist.h json.h json_inttypes.h json_object_private.h linkhash.h bits.h json_c_version.h json_object.h json_tokener.h printbuf.h debug.h json_config.h json_object_iterator.h json_util.h random_seed.h root@ID14337:/usr/local/include/json-c# cd ../../lib root@ID14337:/usr/local/lib# ls libjson-c.a libjson-c.la libjson-c.so libjson-c.so.3 libjson-c.so.3.0.0 pkgconfig python2.6 python2.7 root@ID14337:/usr/local/lib# So any ideas of what could be causing this to fail? Thanks Joao Alves 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 6

Kamailio w/ external routing server
by Eric Babcock 07 Aug '15

07 Aug '15

Hey Everyone, I've been banging my head against the desk for days trying to get this working, and I'm sure it's something simple that I'm not doing, but maybe someone else can add some input. Here is the best way to describe my call flow. Customer -> Kamailio -> Routing Server (302) -> Kamailio -> Vendors (from the 302 header) I have tried to configure dispatch, but that hasn't worked out for me. My routing server does the authentication, LRN lookup, and provides the vendor routes in the sip header. So when I pass the call to the routing server, it has to contain the customer information. This setup has to be stateful so I can write CDRs for billing to MySQL. The reason I tried using dispatch, is because I have multiple routing servers I would like to load balance across or even failover if one goes down. Can anyone help me out, or at least help point me in the right direction? Thanks! -Eric -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-w-external-routing-server-… Sent from the Users mailing list archive at Nabble.com.

3 4

Errors when starting Kamailio with Websocket Module
by Babu Raja 07 Aug '15

07 Aug '15

Hello, I incuded the module for websocket into my Kamailio 4.3 on my Ubuntu 12.04 server environement. For installing the packages and websocket I followed the instructions from the following link: http://www.kamailio.org/wiki/packages/debs http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket After starting the Kamailio I got 26 errors like: can not find command "handle_ruri_alias" or Syntax errors for: route[REGISTRAR] => square brackets.. ($du == "") => parameter "==" unknown.. the whole error log I will paste later.. Can anyone help me?

3 4

RLS Notify sent to wrong transport
by M S 07 Aug '15

07 Aug '15

Hi, I have a Kamailio v4.3.0 with SVN Rev. c6aa95 running an RLS based presence setup. Everything works fine when end-user has UDP transport, however, if user has TCP or TLS transport then i aggregated NOTIFY sent by RLS gives send error, -- WARNING: <core> [forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:X.X.X.X:5061, to udp:Y.Y.Y.Y:12345) -- (X.X.X.X is server IP and Y.Y.Y.Y is end user IP). Not sure if it is configuration issue or some bug, so posting it to both mailing list. Please help. Thank you.

2 8

Kamailio 4.3.1 and Debian 7
by Mathieu Boyer 07 Aug '15

07 Aug '15

Hello, i've got big issues since upgrade of Debian packages (debian repository http://deb.kamailio.org/kamailio wheezy)<http://deb.kamailio.org/kamailio> on 4.3.1. Started with rr.so can't load without outbound.so, location.so find version mismatch (6 vs 8) etc ... Have you got a quick workaround in order to make it run properly ? dpkg -l |grep kamailio ii kamailio 4.3.1+wheezy amd64 very fast and configurable SIP proxy ii kamailio-extra-modules:amd64 4.3.1+wheezy amd64 extra modules for Kamailio ii kamailio-mysql-modules:amd64 4.3.1+wheezy amd64 MySQL database connectivity module for Kamailio ii kamailio-outbound-modules:amd64 4.3.1+wheezy amd64 Outbound module for Kamailio ii kamailio-utils-modules:amd64 4.3.1+wheezy amd64 Provides a set utility functions for Kamailio Error logs : Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4083]: ERROR: dispatcher [dispatcher.c:768]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Unknown column 'server_id' in 'field list' Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: usrloc [udomain.c:424]: preload_udomain(): db_query (1) failed Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: usrloc [ul_mod.c:439]: child_init(): child(1): failed to preload domain 'location' Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [sr_module.c:900]: init_mod_child(): Error while initializing module usrloc (/usr/lib/x86_64-linux-gnu/kamailio/modules/usrloc.so) Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [pt.c:329]: fork_process(): init_child failed for process 1, pid 4085, "udp receiver child=0 sock=XXXX:5063"? Thank you for your help BR Mathieu

3 2

Rtpengine/rtpproxy and zrtp
by Alexandru Covalschi 07 Aug '15

07 Aug '15

Sorry if writing to wrong mailing list, I am very limited to traffic now amd don't know if there is any for rtpproxy/rtpengine. My question is - can they support ZRTP at least in pass-through mode? Will rtpengine fail on trying to recognize unknown SDP fields?

2 2

Kamailio v4.2.6 Released
by Daniel-Constantin Mierla 06 Aug '15

06 Aug '15

Hello, Kamailio SIP Server v4.2.6 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.6 (or to 4.3.x series). For more details about version 4.2.6 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-2-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.1. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

2 3

Kamailio Redis support
by Solutions Solutions 06 Aug '15

06 Aug '15

Hi All, i want to replace MySQL with redis database in kamalio,but in kamalio supports transactions with MySQL ,ie saving data in MySql DB.but while trying to run kamailio with redis ,transaction are not happening,is there any module that i have to enable in kamalio for No SQL databases transaction support.

2 1

Re: [SR-Users] Remote SIP Registration
by Keith 05 Aug '15

05 Aug '15

Hi, I have a carrier who require registration, I have looked at the uac module but can't see how to do this. Anyone point me in the right direction? Thanks Keith

2 10

Difference between corex module's 'alias_subdomains' and 'alias' core parameter
by Iwan Aucamp 05 Aug '15

05 Aug '15

>From the descriptions these seem similar: corex alias_subdomains ( http://kamailio.org/docs/modules/devel/modules/corex.html#idp18017160): Register a domain and all its sub-domains to match the “myself” condition. It can be set many times. Its full format is: 'proto:domain:port', allowing to set restrictions on protocol and port as well. Protocol and port are optional. alias core parameter ( http://www.kamailio.org/wiki/cookbooks/devel/core#alias): Parameter to set alias hostnames for the server. It can be set many times, each value being added in a list to match the hostname when 'myself' is checked. Is the only difference that corex alias_subdomains can include 'proto' ? -- Iwan Aucamp

2 1

Question about how location server_id is supposed to be used.
by Ding Ma 05 Aug '15

05 Aug '15

Looking for some doc that describes how the new columns in 4.3.x location table is supposed to be used. Can find some email discussions about this, but seems no formal doc exists. Would appreciate if someone can provide an explanation. Thanks

3 3

Re: [SR-Users] R: Mutiple branch forking and voicemail
by Daniel-Constantin Mierla 05 Aug '15

05 Aug '15

Indeed usually there are few options for same functionality, good to hear you found a solution. Cheers, Daniel On 03/08/15 14:53, Marino Mileti wrote: > Thanks Daniel, > > I've found another way using a seturi for mediaserver that will be executed > after all branch (using t_load_contacts and t_next_contacts) > In this way I can complete my scenario in the right way > > -----Messaggio originale----- > Da: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Per conto di > Daniel-Constantin Mierla > Inviato: lunedì 3 agosto 2015 12:37 > A: Kamailio (SER) - Users Mailing List > Oggetto: Re: [SR-Users] Mutiple branch forking and voicemail > > Hello, > > On 31/07/15 11:31, Marino Mileti wrote: >> Hello, >> >> I would like to implement a voicemail services but i'm trying >> everything without good results :( >> >> My scenario is: >> >> - A make a parallel forking to B(1001) and C(1002) >> - If timeout, call is routed to voicemail >> >> I've done something like this...very simple >> >> seturi("sip:1001@172.20.40.103"); >> >> append_branch("1002(a)172.20.40.103"); >> route(RELAY) >> >> on failure_route.. >> if (t_check_status("486|408")) { >> $ru = "sip:8000@172.20.40.103:5090"; >> >> route(RELAY); >> >> exit; >> >> } >> >> The scenario is working but on log there are some error... >> t_should_relay_response >> >> I would like that when a branch reach the failure route, all other >> branches will be cancelled..and call is routed to VM. >> Can anyone help me? >> >> > When the failure_route is executed, no branch is active anymore. > > To be able to assist further, you have to send here the errors you get in > the log, to understand what is wrong. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > --- > Questa e-mail è stata controllata per individuare virus con Avast antivirus. > https://www.avast.com/antivirus > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Maintaining order of messages in a sip dialog on forwarding
by Daniel Tryba 05 Aug '15

05 Aug '15

Since switching to 4.2.5 (from 3.x) a customer has problems with a buggy user agent (IMHO). What I am seeing now is: provider kamailio customer INVITE1-> INVITE1-> <-Trying <-Trying <-200 OK <-200 OK ACK1-> INVITE2-> INVITE2-> ACK1-> ACK1 arrives 0.007s before INVITE2, but INVITE2 is send 0.004s before ACK1. It looks like this wasn't happening with Kamailio 3.x, it worked fine then (apparently). Can this reordering be (gracefully) prevented somehow (ignoring the fact that during transport reordering can happen)? A Q&D hack looks to be just usleep the reinvite a little. The endpoint is essentially ignoring the reinvite before (and after) the ACK. After a timeout INVITE2 is canceled and after that INVITE1 will be ended with a BYE from provider. kamailio-customer communication is UDP provider-kamailio is tcp when message >= 1300b. INVITE1 is TCP, INVITE2 and ACK1 are UDP.

2 5

VIA header question.
by Chad 04 Aug '15

04 Aug '15

Hi list, I need a little help, I am a business owner trying to get Kamailio up and running as a SIP load balancer. I hired a Kamailio consultant to help me do so, but Kamailio is not working and I am getting conflicting information. My Kamailio consultant and my VOIP provider are telling me 2 different things and I don't know which one is right. Kamailio sends SIP traffic to the VOIP provider with 2 VIA headers like this (in this order): Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. The VOIP provider says that is incorrect because they are supposed to reply back to the topmost VIA header so they reply to the 10.10.10.254 IP (which is not public) and the call ends. The VOIP provider says Kamailio should send the VIA headers like this instead: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. My Kamailio consultant says the way we are sending it is right and that the VOIP provider is processing the call incorrectly. I read that the SIP proxy is supposed to remove the internal header from the 1st example above based on this RFC: https://tools.ietf.org/html/rfc3261#section-16.7 Item: 3. Via "The proxy removes the topmost Via header field value from the response." If that applies to this situation (which I don't know if it does) then Kamailio should be removing the 10.10.10.254 VIA line and only sending 1 VIA header like this: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Which would sort of make the VOIP provider right in that the topmost VIA line would then be the external IP, but how they said to fix it (reversing the VIA lines) is wrong. Does anyone know what the right answer is here? Please let me know. THANKS for your help. -- ^C

3 3

MSILO: Archiving every message
by SamyGo 03 Aug '15

03 Aug '15

Hi, Tried finding anything as weird as I am trying to do on mailing list but couldn't. The idea is to save a copy of the messages on database. MSILO module can keep the message as long as the destination user is offline, or doesn't support method:MESSAGE or expiry timeout. Any ideas will be appreciated. Best Regards, Sammy

2 2

Re: [SR-Users] Mutiple branch forking and voicemail
by Daniel-Constantin Mierla 03 Aug '15

03 Aug '15

Hello, On 31/07/15 11:31, Marino Mileti wrote: > Hello, > > I would like to implement a voicemail services but i'm trying everything > without good results :( > > My scenario is: > > - A make a parallel forking to B(1001) and C(1002) > - If timeout, call is routed to voicemail > > I've done something like this...very simple > > seturi("sip:1001@172.20.40.103"); > > append_branch("1002(a)172.20.40.103"); > route(RELAY) > > on failure_route.. > if (t_check_status("486|408")) { > $ru = "sip:8000@172.20.40.103:5090"; > > route(RELAY); > > exit; > > } > > The scenario is working but on log there are some error... > t_should_relay_response > > I would like that when a branch reach the failure route, all other branches > will be cancelled..and call is routed to VM. > Can anyone help me? > > When the failure_route is executed, no branch is active anymore. To be able to assist further, you have to send here the errors you get in the log, to understand what is wrong. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

2 1

www.kamailio.org unreachable via ipv6
by Daniel Tryba 03 Aug '15

03 Aug '15

(www.)kamailio.org is unreachable at this time (the whole morning) on 2a00:d60:0:400::2. ipv4 is responsing. Never noticed whether connectivity via ipv6 was working in the past or this is new and misconfigured.

2 1

Announcement: Kamailio is now systemd-rtc-server
by Alex Balashov 03 Aug '15

03 Aug '15

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina,was quick to summarise the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- 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/

5 8

Check if user is registered before reply is routed
by Dmytro Bogovych 01 Aug '15

01 Aug '15

Greetings. Sorry for previous false letter. I want to check if user is registered before reply is routed to its target user agent. Imho it should be like this: onreply_route[MANAGE_REPLY] { if ($rm=="INVITE") { if (registered("location", $ru)) { # Do job - pause for 15 seconds and give chance to reregister } } } But kamailio complains on registered function 0(6274) ERROR: <core> [cfg.y:3295]: yyparse(): misused command registered 0(6274) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1121, column 35: Command cannot be used in the block Documentation confirms - registered() function can be used in REQUEST_ROUTE and FAILURE_ROUTE only. Is it possible to check if user is registered when reply is processed in scripts? Thank you!

4 3

Interpretation of Contact header
by Joachim Büchse 30 Jul '15

30 Jul '15

Good day, I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the" experts about how the requests send to the UAS SHOULD be correctly interpreted. The REGISTER requests/responses look like this (outside of the firewall): Protocol TCP! client port 19091 <-> server port 5060 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch", uri="sip:pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", response="764f371a08d258157a249f8d1b852514" Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall. However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them. Observations: - the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server. - the server does include received="sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error. - after the client received the 200 OK it closes the TCP connection. - the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from). RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) requests from the server to the client. Those are strictly spoken not replies (or are they?). RFC5626 defines that a “proxy” should keep track of the flows over which it received a registration and send requests over the same flow. It is not clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes an “ob” parameter in the Contact field if it whishes further requests over the same flow. Also the RFC mandates a client to add a "reg-id=x" in the Contact field. Both are not the case here, so in short I think RFC5626 should NOT be applied. In which case conecting to the originating port (instead of the Contact port) would be a server error. So in short and if I interpret the RFCs correctly, the client is reachable and should be contacted on Transport: TCP IP: 212.126.160.92 Port: 6717 If anyone who lives and breathes SIP could enlighten me if the UAS is right to call back on 19091 instead of 6717 I would really appreciate it;-)) Best regards, Joachim

3 4

Issue with ACK behind NAT
by Alberto Sagredo 30 Jul '15

30 Jul '15

I have Kamailio in a local network with RTPPROXY and NAT MANAGE I have seen an issue with ACK as show when Asterisk sends ACK to Kamailio, it sends to advertised Address instead to Kamailio IP address. What could i add to solve this? route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails dlg_manage(); } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } BR

2 14

Re: [SR-Users] Memory Issue while accessing $rb (prakash)
by Surendra.P 30 Jul '15

30 Jul '15

Hi Prakash, Specify the more private memory with -M option, while running kamailio Like -M 64 :-) Regards @surendra

1 0

Planning to release v4.2.6
by Daniel-Constantin Mierla 30 Jul '15

30 Jul '15

Hello, I am planning to release v4.2.6 later this week, most likely on Thursday, July 30. If there is anything not reported that should be considered for this minor release of the branch 4.2, write to sr-dev mailing list. 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

Carrierroute and dispatcher
by Nelson Migliaro 29 Jul '15

29 Jul '15

Hello everybody, I am currently using Kamailio in order to separate traffic based on dialed number. Some traffic goes to several Asterisk in a load balance strategy and other traffic goes to other SIP devices. I use carrierroute module in order to route traffic based on destination. I use scan_prefix to classify traffic and rewrite_host to route the traffic. A destination number usually go to one of 4 asterisk server. I do this in order to load balance the traffic. In order to distribute traffic I use prob (0.25 for each destination). Using this configuration, I have to include on carrieroute table a row for each destination node. Since the amount of destination is huge, I have to repeat configuration in tables. It also complicates maintenance. I was thinking of using a combination of carrierroute and dispatcher module. Carrierroute for classify traffic, strip numbers, and rewrite destination host and dispatcher to distribute the traffic over asterisk. Using this configuration I have to indicate asterisk nodes once in dispatcher table once. Also in carrierroute table I have to indicate each prefix once. I was thinking in something like this. if(!cr_route("carrier", "domain", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); drop; } if(!ds_select_dst("1", "4")) { sl_send_reply("403", "Not allowed"); drop; } t_relay; Does it make sense, or am I complicating everything and there is a magic way to achieve this. Thank you.

2 1

Re: [SR-Users] Memory Issue while accessing $rb
by Daniel-Constantin Mierla 29 Jul '15

29 Jul '15

Hello, you have to increase the private memory -- see -M command line parameter -- as well as pv_buffer_size core parameter -- see core cookbook in the wiki. Cheers, Daniel On 29/07/15 13:37, prakash wrote: > > Hello, > > > > I am getting this Error: > > > > *ERROR: <core> [rvalue.c:1706]: rval_str_add2(): rv eval out of memory* > > *ERROR: <core> [lvalue.c:389]: lval_assign(): rval expression > evaluation failed (881,17-881,29)* > > > > When I tried access *$rb* when sending large amount of data(*nearly > 4MB*) as a XML body. > > How can I sort out this issue? Is there anything to be configured > properly? > > > > NOTE: > > Version that I’m using kamailio 4.1.5 > > > > Thanks and Regards, > > Prakash P > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Kamailio no audio
by goReeves Admin 29 Jul '15

29 Jul '15

Hi - I just installed Kamailio 4.3.0 on Ubuntu 14.04.2. I can connect using jitsi on windows and csipcimple on the android and do text messaging. However, when I try to do audio callis it seems that the mic is working but the audio out is not functioning. I used the tutorial to do the install so everything in the config files is pretty vanilla except for the database name, use and password. I also installed siremis and added my sip.xxxxx.com as a gateway. On the Ubuntu image I use VPN to a dedicated IP address on which all ports are open. I run web and cloud servers on this machine as well and they all work just fine. I am pretty new to this SIP stuff so it is probably my issue. Here is the network configuration: root@xxxxxxxx-VM:/var/www/siremis-4.2.0# ifconfig eth0 Link encap:Ethernet HWaddr xx:xx:xx:xx:xx:xx inet addr:192.168.x.x Bcast:192.168.x.255 Mask:255.255.255.0 inet6 addr: fe80::20c:29ff:fe84:447a/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1579455 errors:0 dropped:0 overruns:0 frame:0 TX packets:1444329 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:346579445 (346.5 MB) TX bytes:625010061 (625.0 MB) lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:65536 Metric:1 RX packets:98232 errors:0 dropped:0 overruns:0 frame:0 TX packets:98232 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:262176392 (262.1 MB) TX bytes:262176392 (262.1 MB) ppp0 Link encap:Point-to-Point Protocol inet addr:xxx.xxx.xxx.xxx P-t-P:xxx.xxx.xxx.xxx Mask:255.255.255.255 UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1400 Metric:1 RX packets:1105547 errors:0 dropped:0 overruns:0 frame:0 TX packets:1228003 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:3 RX bytes:184558026 (184.5 MB) TX bytes:235399612 (235.3 MB) -- Web Administrator <mail:webadmin@goreeves.net> goreeves <https://www.goreeves.net>

2 2

Memory Issue while accessing $rb
by prakash 29 Jul '15

29 Jul '15

Hello, I am getting this Error: ERROR: <core> [rvalue.c:1706]: rval_str_add2(): rv eval out of memory ERROR: <core> [lvalue.c:389]: lval_assign(): rval expression evaluation failed (881,17-881,29) When I tried access $rb when sending large amount of data(nearly 4MB) as a XML body. How can I sort out this issue? Is there anything to be configured properly? NOTE: Version that I'm using kamailio 4.1.5 Thanks and Regards, Prakash P

1 0

double rr and $fs
by Asgaroth 29 Jul '15

29 Jul '15

Hi All, I have kamailio setup and listening on logical interfaces (for failover purposes) and therefore need to force the sending socket on initial messages. We are testing sips+tls at the moment and what we are trying to achive is that the TLS connection occurs from the UAC to the edge proxy (also responsible for the domain) and then force UDP for internal communications to services. What I am comming up against is that when I force the sending socket on the internal interface to UDP, the record route header still shows up as a "sips" request. Then, at the presense server, when generating the NOTIFY it attempts to send it to the proxy using TLS. My understanding was that the record route would set the uri something like "sip:internal_ip" when forcing the sending socket to be "udp:internal_ip". Is my understanding correct, or am I doing something wrong here? An example subscribe message follows, the top most record-route header, from my understanding, should read sip:internal_ip, not sips:internal_ip. Is this the expected behaviour of double rr with these types of messages? How could I ensure that the top most record-route would always be a sip uri (never sips). SUBSCRIBE sips:subscriber@domain.com:5061 SIP/2.0. Record-Route: <sips:proxy_internal_ip;r2=on;lr;ftag=594055226>. Record-Route: <sips:proxy_external_ip:5061;transport=tls;r2=on;lr;ftag=594055226>. Max-Breadth: 60. Via: SIP/2.0/UDP proxy_internal_ip;branch=z9hG4bK6c9.d7dc4801e872ce9fb5730f9e09f1889e.0;i=1. Via: SIP/2.0/TLS 172.16.0.121:5060;rport=33273;received=78.143.152.30;branch=z9hG4bK708ea7ba. Max-Forwards: 69. Call-ID: 17212ff4-4421321c8feafd63bf800080f0808080@KX-HDV230X. From: <sips:subscriber@domain.com>;tag=594055226. To: <sips:subscriber@domain.com>. CSeq: 2 SUBSCRIBE. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER. Accept: application/dialog-info+xml,application/rlmi+xml,multipart/related. Contact: <sips:subscriber@172.16.0.121:5060;alias=78.143.152.30~33273~3>. Expires: 3600. Event: dialog. Any thoughts, tips, tricks would be greatly appreciated. Cheers

2 2

Kamailio - Missing INVITEs
by Ali Taher 28 Jul '15

28 Jul '15

Hello, I connected Kamailio to a postgresql database (through sqlops module) and fetching result from a table containing around 300K records , where time processing of the single query is taking around 12 ms. I noticed that on huge load (around 200 - 300 calls per second) Kamailio is not processing all INVITE packets. For instance , I initialized a counter in Kamailio that increments on every INVITE , and on the other hand I counted the number of INVITEs found in Wireshark. I got a big difference (from 30K INVITES in Kamailio , I got 15K in Kamailio), though no errors were raised in kamailio log. How can I proceed in order to detect the problem? :( Any help would be appreciated. Regards, Ali

2 1

RR module - Fail detecting strict routing
by Antonio Reale 28 Jul '15

28 Jul '15

Hi all, I have the following scenario: U1 (caller) ---> P1 (192.168.0.245, kamailio 4.3, loose-router) ----> P2 ----> (192.168.0.101, strict router) ----> .... ----> U2 (called) When U2 answers the call, at P1 arrives the 200 OK with: Record-Route: <sip:0000000011@192.168.0.101;transport=udp;pgw-call=call-28d2e> Record-Route: <sip:192.168.0.245;lr=on> The problem is that the ACK from U1 is forwarded from kamailio to P2, as if P2 is a loose router. P2 drops the ACK message. From kamailio logs I see: Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:88]: is_preloaded(): is_preloaded: No Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:783]: after_loose(): Topmost route URI: 'sip:192.168.0.245;lr=on' is me Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:869]: after_loose(): URI to be processed: 'sip:0000000011@192.168.0.101;transport=udp;pgw-call=call-289ce' Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:878]: after_loose(): Next URI is a loose router Why after_loose function considers the next URI a loose router? The Route HF in the ACK with the URI of the next hop does not contain ;lr . It seems that the function is_strict fails detecting the strict router. Thanks. Regards. Antonio

2 7

CNXCC lock
by Kelvin Chua 28 Jul '15

28 Jul '15

after a call exhausted credits, the next call will fail with a timeout. found out it gets stuck in cnxcc_mod.c line 1137 static call_t *__alloc_new_call_by_money(credit_data_t *credit_data, struct sip_msg *msg, double credit, double cost_per_second, int initial_pulse, int final_pulse) { call_t *call = NULL; lock_get(&credit_data->lock); ... the next lines will never be executed Kelvin Chua

2 2

SIPCapture and Message Queues
by Grant Bagdasarian 27 Jul '15

27 Jul '15

Hello, Would it be possible to send the messages captured by the sipcapture module to a message queue instead of inserted directly into a database? Be it DMQ or SQL Server Service Broker Queues, etc. Or maybe another approach, executing a stored procedure instead of an insert statement? I couldn't find a way to accomplish this, but maybe someone has a brilliant idea how to accomplish any of the above. Regards, Grant

2 2

Problem Storing USRLOC Additional Attributes
by Luca Mularoni 27 Jul '15

27 Jul '15

Hi All, I'm using kamailio 4.2.5 (compiled from source and aligned up to commit [00a36ae07c587992d6486277ce2e9ae6c009685f]). In order to enrich users data collected during registering phase, I would like to store them in USRLOC Additional Attributes through the use of 'xavp_contact' structure, but I'm having problems storing them. Below relevant chunks of config: ... # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("usrloc", "xavp_contact", "ulvals") ... route[REGISTRAR] { ... # Just an example $xavp(ulvals=>A) = "A"; $xavp(ulvals[0]=>B) = "B"; $xavp(ulvals[0]=>C) = "C"; $xavp(ulvals[0]=>D) = "D"; ... if (!save("location")) sl_reply_error(); exit; } route[LOCATION] { ... if(!lookup("location") { # Handle Failure ... } # Print USRLOC Additional Attributes # Just an Example $var(i) = 0; $var(N) = $branch(count); while($var(i)<=$var(N)) { xlog("LOC - [$var(i)] - A [$xavp(ulvals[$var(i)]=>A)]\n"); xlog("LOC - [$var(i)] - B [$xavp(ulvals[$var(i)]=>B)]\n"); xlog("LOC - [$var(i)] - C [$xavp(ulvals[$var(i)]=>C)]\n"); xlog("LOC - [$var(i)] - D [$xavp(ulvals[$var(i)]=>D)]\n"); $var(i) = $var(i) + 1; } ... } And this is the relevant chunk of syslog when a local subscriber is found in 'location' table: ... LOC - [0] - A [A] LOC - [0] - B [<null>] LOC - [0] - C [<null>] LOC - [0] - D [D] ... Summing up, it seems that when xavp structure is stored in memory, just first and last elementsare inserted. In real config I'm using more than 4 attributes and behaviour is the same: just first and last are stored. I've tried also with modparam("usrloc", "db_mode", 0) but the problem remains. Where I'm wrong in using this structure? Or it could be a bug? Thank You in advance for your help Luca Mularoni

2 11

Check if user is registered after receiving reply
by Dmytro Bogovych 27 Jul '15

27 Jul '15

Greetings. I want to check if user is registered before kamailio will try to route reply to its user agent. IMHO it should be: onreply_route[MANAGE_REPLY] {

2 1

Re: [SR-Users] FROM number based on destination number
by Keith 27 Jul '15

27 Jul '15

Hi, I have a requirement to change the FROM number to a different number depending on the destination number. For example if I dial 01789 from 0800 I want to change the 0800 from number to a 01789 number. Ideally I want to be able to do a db lookup on the destination number to then set the from number. Thanks Keith

1 0

Changing acc_cdrs time and duration columns from varchar to int/float
by mayamatakeshi 27 Jul '15

27 Jul '15

Hello, I am planning to enable acc cdr in my kamailio 4.1 server. I am testing in my lab and it is working fine. However, when I check the table definition I see this: mysql> describe acc_cdrs; +------------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +------------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | start_time | varchar(32) | NO | MUL | | | | end_time | varchar(32) | NO | | | | | duration | varchar(32) | NO | | | | +------------+------------------+------+-----+---------+----------------+ 4 rows in set (0.00 sec) The *_time and duration columns are defined as varchar. Because of this, i need to perform a cast when doing queries: select * from acc_cdrs where cast(start_time as unsigned) >= 1437120622; which would cause extra processing in the db server. So I experimented changing these columns to "int unsigned" and to "float unsigned" and it worked for both without problems (records properly added to acc_cdrs and no errors in kamailio.log): alter table acc_cdrs modify start_time int unsigned; alter table acc_cdrs modify end_time int unsigned; alter table acc_cdrs modify duration int unsigned; alter table acc_cdrs modify start_time float unsigned; alter table acc_cdrs modify end_time float unsigned; alter table acc_cdrs modify duration float unsigned; So I am thinking in doing this in my production server but I would like to know if there is any potential risk of doing this (as I am not sure why it was decided for these fields to be varchar instead of a numeric type). Regards, Takeshi

2 3

New SIP INVITE from UAS (new dialogue)
by Joao Alves 27 Jul '15

27 Jul '15

Hi, What would be the best way to generate and send an INVITE message generated by the Kamailio server. This would be for a new dialog session, UAS initiated (e.g. to make a click 2 call like service). I've tried the $uac_req(method)="INVITE"; uac_req_send(); but didn't work. I gather something would be needed first to open the dialogue before doing it but couldn't find it on the dialogue module. Thanks, Joao Alves 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

4 10

Issues sending SIP UPDATE inside an existing dialogue.
by Joao Alves 24 Jul '15

24 Jul '15

Hi, I need to send an SIP UPDATE inside an existing dialogue (in order to update the SDP offer). I've been using the following uac functions from the command file. $uac_req(method)="UPDATE"; $uac_req(ruri)=$shv(userAcontact); $uac_req(turi)=$tu; $uac_req(furi)=$fu; $uac_req(callid)=$ci; $uac_req(hdrs)="Content-Type: application/sdp\r\n"; $uac_req(body)=$var(sdp); # uac_restore_from(); uac_req_send(); As a result, the SIP UPDATE is being sent but since the original From Tag is not being used (i.e. and a new tag is generated), the SIP client (sipML5) does not recognize it as part of the existing dialogue. Attach is a pcap of the test made. Subsequently, I've also tried to recover the from using uac_restore_from() function but got the following an error message. DEBUG: uac [replace.c:511]: restore_uri(): getting 'vsf' Route param DEBUG: uac [replace.c:515]: restore_uri(): route param 'vsf' not found Thanks, 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

1 0

Installation issues
by ngn zone 24 Jul '15

24 Jul '15

>Hello all, I am new to Kamailio. I Tried installing Kamailio 4.0 and I had the following error. I will appreciate any help >ERROR: could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL'

2 1

dialog keepalive timeout - which parameter rules it?
by Andrey Utkin 24 Jul '15

24 Jul '15

What happens if dialog parties don't respond to keepalive requests sent by dialog module? It looks it is not coded in modules/dialog, I guess it can be set in t_uac_prepare() from modules/tm/uac.c. Please explain what timeout or lifetime parameters rule - at which delay the unanswered keepalive request sent by Kamailio within dialog triggers dialog termination. Thanks a lot. -- Andrey Utkin

2 2

dmq_usrloc replication and max_contacts
by Asgaroth 24 Jul '15

24 Jul '15

Hi All, I have a situation where I have 2 devices registering with the same AOR, I have the registrar module's max_contacts parameter set to 1 and I use the 0x04 flag on the save function. When I use dmq_usrloc for replication, I can see 2 contacts registered for the AOR on the "client" nodes whereas the node where the actual registration took place only has 1 contact for the AOR. Is this expected behaviour of the dmq_usrloc module in terms of registration or have a come accross another bug? Thanks

2 5

"kamctl ul show"/xmlrpc ul.lookup AOR not found for valid registration
by Asgaroth 24 Jul '15

24 Jul '15

Hi All, I have a situation where a device has successfully registered and its location saved/stored in memory. I can see all the stored information when I issue a "kamctl ul show" However, if I perform a kamctl ul show subscriber@domain, where the 'subscriber@domain' is copied/pasted from the output of the "kamctl ul show" command, it results in an AOR not found error. However, with other AOR's on the same registrar, the "kamctl ul show subscriber@domain" works. I checked the usernames for the failed commands, and I dont see anything obvious, they are a combination of lower/upper-case leters and digits. String length is around 24 characters for the username (the username table for the subscriber is 64 characters). I'm not sure what elese to check to see why the AOR lookup for these particular clients are failing when they are indeed registered. All other functions around the lookup seem to work, for example, a location lookup for these AOR's are working. I wonder if I have come accross an issue with the kamctl function and/or the equivilient xmlrpc ul.lookup method. Any thoughts on this would be greatly appreciated. Thanks

1 3

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

2 1

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

2 2

Error: make modules include_modules ="db_mysql" cfg
by kumar m 16 Jul '15

16 Jul '15

I am a newbie trying out kamailio *kamailio version: *4.3.0 *Operating system:* CentOS 7 Following the instructions provided at http://www.kamailio.org/wiki/install/4.3.x/git While I was trying to include the module "db_mysql", I get the following error [root@localhost kamailio]# make modules include_modules="db_mysql dialplan app_mono" cfg Makefile:56: modules.lst: No such file or directory target architecture <x86_64>, host architecture <x86_64> saving modules list... target architecture <x86_64>, host architecture <x86_64> which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) make --no-print-directory -wC ../.. cfg-defs target architecture <x86_64>, host architecture <x86_64> making config... which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) CC (gcc) [M db_mysql.so] km_dbase.o km_dbase.c:36:25: fatal error: mysql/mysql.h: No such file or directory #include <mysql/mysql.h> ^ compilation terminated. make[1]: *** [km_dbase.o] Error 1 make: *** [modules] Error 1 Please help me in fixing the error

2 1

Unclear documentation for "priority" field in Dispatcher module
by Sven Neuhaus 16 Jul '15

16 Jul '15

Hello, the documentation doesn't clearly state if a larger value for the priority field means that it gets used first before entries with smaller priority values are used - or vice versa. It currently states: “8” - select destination sorted by priority attribute value (serial forking ordered by priority). and priority: sets the priority in destination list (based on it is done the initial ordering inside the set) This is just ambiguous. It should clearly say wether the list is processed in ascending or descending order for the priority value when using algorithm 8 (serial forking aka failover). Regards, -Sven Neuhaus

2 1

Issue handling SRTP and RTP with rtpproxy and rtpengine
by Alberto Sagredo 15 Jul '15

15 Jul '15

Hi Im using an Asterisk cluster behind Kamailio and some services are running on Asterisk. Traffic always must go to Asterisk so RTPPROXY and RTPNGINE do the work and send traffic to Asterisk and traffic from Asterisk go to RTP Proxy I have been able to get rtpproxy working and later tried RTPengie to be able to manage SRTP I have been able to make SRTP To RTP to Asterisk Im able to call between RTP using rtpproxy as recording is a must and rtpengine does not record yet. But im not able to call between SRTP extensions, i understand also SRTP to RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine trasncode) I have this code on NATMANAGE but i do not know what could be wrong to handle SRTP Any idea? Thanks in advance if(src_ip==ASTERISK1_LAN || src_ip==ASTERISK2_LAN ){ if (sdp_get_line_startswith("$avp(mline)", "m=")) { if ($avp(mline) =~ "AVP") { add_rr_param(";rtp=AVP-kamtoast"); set_rtp_proxy_set("1"); rtpproxy_manage("rwei"); } if ($avp(mline) =~ "SAVP") { add_rr_param(";rtp=SAVP-kamtoast"); rtpengine_manage("trust-address direction=external replace-origin replace-session-connection ICE=remove RTP AVP"); } } } else{ if (sdp_get_line_startswith("$avp(mline)", "m=")) { if ($avp(mline) =~ "AVP") { add_rr_param(";rtp=AVP-asttokam"); set_rtp_proxy_set("1"); rtpproxy_manage("rwie"); } if ($avp(mline) =~ "SAVP") { add_rr_param(";rtp=SAVP-asttokam"); rtpengine_manage("trust-address direction=external replace-origin replace-session-connection ICE=remove "); } } } If you need any more info let me know.

4 12

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

15 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.

1 0

ISUP decode and store to be able to append_body_part() later
by andre second 15 Jul '15

15 Jul '15

Hi, I am trying to extract and save ISUP from SIP-I packet for a later use. The problem is that data is stored in binary format and needs decoding before saving to a variable. Can Kamailio perform any decoding so that later I can use append_body_part() to append it back to a multipart message? Thanks,Andrei

1 0

License for SIP-Router server when used as part of a commercial product development
by Mahesh Kumaraguru 15 Jul '15

15 Jul '15

We wish to use SIP-Router server for our commercial product development. We plan to develop SIP clients for Android, iOS, Windows phone and PC browser to have voice and video calls. Please let us know the licensing terms under which SIP-Router server is available, especially in the context of us using it for our commercial product. Many Thanks and Best Regards, Mahesh Kumaraguru +(91)-942-417-4471 +(91)-835-782-9220 mahesh.kumaraguru(a)vectramind.com<mailto:mahesh.kumaraguru@vectramind.com> http://www.vectramind.com/

1 0

License for iptel SIP server when used as part of a commercial product development
by Mahesh Kumaraguru 15 Jul '15

15 Jul '15

We wish to use iptel SIP server for our commercial product development. We plan to develop SIP clients for Android, iOS, Windows phone and PC browser to have voice and video calls. Please let us know the licensing terms under which iptel SIP server is available, especially in the context of us using it for our commercial product. Many Thanks and Best Regards, Mahesh Kumaraguru +(91)-942-417-4471 +(91)-835-782-9220 mahesh.kumaraguru(a)vectramind.com<mailto:mahesh.kumaraguru@vectramind.com> http://www.vectramind.com/

1 0

Issues with Yealink SDP and sdp_with_transport()
by Alberto Sagredo 14 Jul '15

14 Jul '15

Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is send SAVP or not to divert to and rtp proxy or rtpengine, as you know rtpproxy supports recording and rtpengine does not yet. So when using if(sdp_with_transport("RTP/SAVP")) with Grandstream Phones all worked fine, but when configuring Optional or Compulsory SRTP in Yealink it seems to do not detect i have seen that crypto lines are not in the final SDP but do not know if thats the reason Did you have a similar issue with Yealink? If i could get traces in anyway to help let me know. BR Alberto INVITE sip:212@10.0.1.34:15060 SIP/2.0. Record-Route: <sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes>. Record-Route: <sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes>. Via: SIP/2.0/UDP x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1. Via: SIP/2.0/TLS 10.0.1.111:11891 ;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360. From: "214" <sip:214@1x.x.x.x:8001>;tag=1072578853. To: <sip:212@x.x.x.x:8001>. Call-ID: 0_1310998066(a)10.0.1.111. CSeq: 2 INVITE. Contact: <sip:214@83.x.x.x:11891;transport=TLS>. Content-Type: application/sdp. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. Max-Forwards: 69. User-Agent: Yealink SIP-T21P_E2 52.80.0.3. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 553. . v=0. o=- 20143 20143 IN IP4 x.x.x.x. s=SDP data. c=IN IP4 x.x.x.x t=0 0. m=audio 8530 RTP/AVP 0 8 18 101. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3. a=crypto:3 F8_128_HMAC_SHA1_80 inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=nortpproxy:yes

2 2

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users