- sr-users - kamailio.org

Problem restoring To and From (uac_replace)
by Duarte Rocha 24 Sep '19

24 Sep '19

Greetings, I'm getting some errors with the auto restore of From and To from UAC Module. After the restore, the URI has lots garbage and it's not like the original. I think the problem lies on the way the client sends the Route Header to me, but i'm not sure. When Kamailio sends the INVITE Record-Route are like this : Record-Route: <sip:xxx.xxx.xxx.xxx;................> Record-Route: <sip:aaa.bbb.aaa.bbb;................> However, when my client sends a BYE to finish the call, the Route are on this mode : Route: <sip:aaa.bbb.aaa.bbb;................>,<sip:xxx.xxx.xxx.xxx;................> The module's documentation says this : "One uses the rr module and appends to the Record-Route header a parameter containing some strings from which the original and new URI can be computed. The problem with this mode is that it relies on the fact the parties will send the Route exactly as it was received. In case there is a change, the resulting URIs will not be correct." Can this change by the client be the cause of this issue? Is there anything i can do to avoid this? Best Regards, Duarte Rocha

1 0

Can I load balance Kamailio with Kamailio?
by Ne Mo 24 Sep '19

24 Sep '19

3 2

Parsing json file on kamailio.cfg file
by Melek Oktay 24 Sep '19

24 Sep '19

Hi, I am trying to monitor *uac *channels status periodically in kamailio config file. I call uac.reg_dump rpc command and result will be inside in $var(uac_channel) variable jsonrpc_exec('{"jsonrpc": "2.0", "method": "uac.reg_dump"}'); $var(uac_channel) = $jsonrpl(body); it returns me json formatted result like below, it has a several channels info and their status *flags* {"jsonrpc":"2.0", "result":[ {"l_uuid":"YY","l_username":"YY","l_domain":"YY","r_username":"YY","., " *flags*":16 }, {"l_uuid":"XX","l_username":"XX","l_domain":"XX","r_username":"XX",,... " *flags*":1 }}]} I could not parse this result on kamailio.cfg file ! I want to learn channel that has a 16 value in flag attribute for example !! Would you help me to figure out how I am gonna parse this json file please ?

2 2

qm_find_free() Free fragment not found, called from xcap_server, no more pkg
by Cody Herzog 23 Sep '19

23 Sep '19

Hello. Recently, for the first time, we experienced an apparent memory issue in our production environment. Here's an example of the relevant log messages: Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: <core> [core/mem/q_malloc.c:286]: qm_find_free(): qm_find_free(0x7fd8bbcfc010, 134648); Free fragment not found! Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fd8bbcfc010, 134648) called from xcap_server: xcap_server.c: ki_xcaps_put(549), module: xcap_server; Free fragment not found! Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: xcap_server [xcap_server.c:552]: ki_xcaps_put(): no more pkg Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2252]: ERROR: app_perl [kamailioxs.xs:487]: XS_Kamailio_log(): 500 Server error The failed operation was the XCAP server module trying to generate a very large RLS contact list for a user. The issue only impacted users with very large lists, as though a large contiguous block of memory could not be found, whereas other smaller allocations continued to work fine. We believe the requested allocation was around 112 KB in size. The server had been up for 14 days. We were able to work around the issue temporarily by just restarting the Kamailio service. It's unusual, because our production server is often up for months, and we've never seen this issue before. The load on production is increasing slowly due to an increased concurrent user count, so that might be related. Before restarting the service on production, we captured the output of the following commands: kamctl stats shmem kamcmd mod.stats all shm kamcmd pkg.stats kamcmd mod.stats all pkg Here's the shared mem output: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 27240", "shmem:free_size = 447203296", "shmem:max_used_size = 116175576", "shmem:real_used_size = 89667616", "shmem:total_size = 536870912", "shmem:used_size = 68824240" ], "id": 6934 } Here's the pkg output for the particular PID which was throwing the errors: { entry: 34 pid: 2302 rank: 14 used: 2415864 free: 4949688 real_used: 3438920 total_size: 8388608 total_frags: 1951 } We didn't see anything obvious in the stats output which explains the issue. We've been trying to reproduce the issue in a dev environment using simulated higher than production load running for many days, but so far we've had no luck. We've been monitoring memory stats over time, but we don't see any obvious leaks or issues. We've searched various past threads, but didn't find any obvious answers. Here are some of the documents and the threads we've been reading: https://www.kamailio.org/wiki/tutorials/troubleshooting/memory https://www.kamailio.org/wiki/cookbooks/3.3.x/core#mem_join https://sr-users.sip-router.narkive.com/3TEDs3ga/tcp-free-fragment-not-found https://lists.kamailio.org/pipermail/sr-users/2012-June/073552.html https://lists.kamailio.org/pipermail/sr-users/2017-February/096132.html https://lists.kamailio.org/pipermail/sr-users/2017-September/098607.html https://github.com/kamailio/kamailio/issues/1001 https://lists.kamailio.org/pipermail/sr-users/2016-April/092592.html https://lists.kamailio.org/pipermail/sr-users/2010-July/064832.html Regarding our Kamailio version and build options, here's the output of 'kamailio -v': ---- version: kamailio 5.1.0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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: unknown compiled on 20:07:31 Jan 4 2018 with gcc 5.4.0 ---- We are running on a 64-bit Ubuntu Server virtual machine. Any help would be greatly appreciated. Thanks very much.

3 6

Using static entries with usrloc
by George Diamantopoulos 23 Sep '19

23 Sep '19

Hello, Normally I wouldn't be asking this but I was surprised to find no reference to anyone doing this online. I need to have a few AoRs with static entries in kami's location table, which will always be reachable at their contacts without ever having to register with Kamailio. Am I right in thinking that merely INSERTing respective rows in kamailio's location DB backend will do the trick? Is there anything I need to worry about, like contacts being periodically purged? If yes, how do I prevent that from happening? And will kamailio always query the DB when performing location lookup or do I need to restart upon inserting new records (to purge caches, for example)? Lastly, I'm not sure how to treat some fields present in usrloc tables like ruid, expires (can I set this to NULL?), callid and cseq (which is irrelevant since there's no REGISTER), flags, cflags and methods. Thanks, George

5 7

xtrange error but those rules that already work, some bug?
by PICCORO McKAY Lenz 23 Sep '19

23 Sep '19

i have many xtrange errors in the log.. but the code seems as i guess completed each rule without error. In the kamailio i have that error by example: Sep 23 13:38:29 ser-server-02 kamailio[2847]: 4(2855) ERROR: *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio-devel.cfg] l=692 a=25 n=save that line represents a supposed error in the following code, BUT that code seem work.. kamailio it’s working perfectly.. so i’m worry about it, what it happened? # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } why that happened? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

kamailio run in docker not working
by Gaurav Bmotra 23 Sep '19

23 Sep '19

hi i m trying to run kamailio in docker this kamailio/kamailio-ci:master-alpine but inside the container kamailio not running if i rung this / # /etc/init.d/kamailio status sh: /etc/init.d/kamailio: not found plz help me thanks -- *Regards:* Gaurav Kumar

2 3

rate-limiting enabled on kamailio.org
by Henning Westerholt 22 Sep '19

22 Sep '19

Hello, just to inform you that rate-limiting on kamailio.org web server has been enabled. The motivation is to throttle robots and other automated services to not overload the server. If you notice any new errors in accessing the service (like page not found, or a 429 Too Many Requests) let us know. Cheers, Henning -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

1 0

TLS in-dialog set_forward_no_connect()and upstream TLS LCR gateway
by Anthony Joseph Messina 21 Sep '19

21 Sep '19

In preparation for the 5.3 release, I've been testing the following configuration change for TCP/TLS connections: https://github.com/kamailio/kamailio/commit/ 8bba208fe6ae7ccb4c92362b8c33f1530b9f56da route[REQINIT] { # no connect for sending replies set_reply_no_connect(); if(has_totag()) { # no connect for requests within dialog set_forward_no_connect(); } This change creates issues when a UAC TLS INVITE routes to an upstream gateway using TLS to port 5061 (via the LCR module). Kamailio sends the initial outbound TLS connection from a local ephemeral port. The TCPOPS tcp_keepalive_enable function issues keepalives from the local ephemeral port to the gateway port 5061: https://kamailio.org/docs/modules/stable/modules/ tcpops#tcpops.f.tcp_keepalive_enable Even so, the TLS connection eventually times out, after which in-dialog requests from the UAC are no longer able to reach the upstream gateway. ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failed WARNING: tm [t_fwd.c:1570]: t_send_branch(): sending request on branch 0 failed ERROR: sl [sl_funcs.c:372]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) I figure I must be doing something wrong with my TCPOPS here. Is a TLS connection to an upstream gateway supposed to be maintained throughout the duration of a call? -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 6

Info: initialization of libssl 1.1.+ with PTHREAD_PROCESS_SHARED attribute for pthread mutex and rwlock
by Daniel-Constantin Mierla 21 Sep '19

21 Sep '19

Hello, after couple of months waiting for feedback about the workaround with the preloaded shared library to set the PTHREAD_PROCESS_SHARED attribute for pthread mutex/rwlock in order to avoid deadlocks and as it seems to be the solution, I just pushed a few commits to move that code in the core of kamailio. It this way the initialization with the PTHREAD_PROCESS_SHARED attribute is done by default when libssl v1.1 is detected at compile time, without the need of using the approach with a preloaded library. The shared library is still kept in the tls module: * https://github.com/kamailio/kamailio/tree/master/src/modules/tls/utils/open… Probably is going to be removed in the near future. I am asking the community members that use Kamailio on systems with libssl v1.1+ to test the master branch and give feedback. The master branch is anyhow in testing phase for preparing the next major release v5.3.0, so it will help for that as well. If no issue is going to be reported soon, the commits will be backported to stable branches, to re-enable smooth use of Kamailio with libssl v1.1+. When libssl v1.1+ is detected, the core links with pthread, but that was already present in the Makefiles.defs for different compile options. The pthread more or less a default library on linuxes/unixes, so this should not introduce any major dependency to the Kamailio core. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

6 21

Possible locking issue in usrloc module in file udomain.c
by Rupesh Kumar 18 Sep '19

18 Sep '19

Hi , Currently I am running kamailio 4.2 and now upgrading to kamailio 5.2 and was just going thru the changes. In the file modules/usrloc/udomain.c inside the function free_udomain( ) , it appears that we are not protecting via lock_ulslot and unlock_ulslot doing free. Could it cause some race condition ? May be its intentional but going thru the github history its there since day 1. I checked all branches in kamailio 4.x and we have the protection. In 4.x void free_udomain(udomain_t* _d) { int i; if (_d->table) { for(i = 0; i < _d->size; i++) { lock_ulslot(_d, i); deinit_slot(_d->table + i); unlock_ulslot(_d, i); } shm_free(_d->table); } shm_free(_d); } In 5.x void free_udomain(udomain_t* _d) { int i; if (_d->table) { for(i = 0; i < _d->size; i++) { deinit_slot(_d->table + i); } shm_free(_d->table); } shm_free(_d); }

2 2

How to distinguish is from Asterisk or from the client
by przeqpiciel 18 Sep '19

18 Sep '19

Hi, I looking information how to distinguish whether the packet is directed to the Astersik (my platform) or to the customer (softphone / sip client ).

5 5

NAT problem
by Arnout Van Den Kieboom 18 Sep '19

18 Sep '19

Hi, First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one. I started using kamailio in a test environment about a week ago. for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio. I have installed rtpproxy and it works nicely. There is just one situation where it fails: Carrier --> FW (NAT) Kamailio --> asterisk There is only one way audio (from asterisk to carrier) but not from carrier to asterisk. I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy... Thanks Pemp

4 7

app_lua: invalid Lua environment attributes or parameters
by Yuriy Gorlichenko 18 Sep '19

18 Sep '19

Hi! Im using kamailio 5.1.8 with app_lua and getting this message during startup of kamailio. Looks like I forgot to add something at the module setup, but cant figure out what is wrong. My setup of app_lua is: modparam("app_lua", "load", "/etc/kamailio/extended/main.lua") modparam("app_lua","reload",1) modparam("app_lua","register","textops") app_lua [app_lua_sr.c:1599]: sr_kemi_lua_exec_func_ex(): invalid Lua environment attributes or parameters starting kamailio service with kamailio -f /etc/kamailio/extended/main.lua Also here is little bit confusing me that in case of use "KSR" everywhere I'm getting message about "sr"... Thx in advice

2 5

dedicated bearer - Kamailio settings
by Jack R 18 Sep '19

18 Sep '19

Hi, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu. I have done the attached (image) settings to create dedicated bearer in nextepc as advised by this link: https://github.com/open5gs/nextepc/issues/180 Do I need to do some setting in Kamailio server side (maybe in pcscf cfg files etc.to take some information from PCRF) to make dedicated bearer work end to end or that is all (just configure the new apn in the phone) rest will be taken care itself? Thanks and Regards, Jack.

2 1

core dump if silo table > ~5000 records
by David Villasmil 18 Sep '19

18 Sep '19

Hello guys, i've got a kamailo: version: kamailio 5.0.0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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: unknown compiled on 19:50:58 Mar 28 2017 with gcc 4.9.2 which was core dumping every 20 seconds and we had no idea why. We emptied out the silo table, which had: mysql> select count(*) from silobackup; +----------+ | count(*) | +----------+ | 5240 | +----------+ records. Once we truncated it, kamailio started working again. Is there a limit to how many records msilo can handle? I will look into the data in the table, but in case there's a limit, i'd like to know. core available on request. Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Kamailio+B2BUA on IPv4-only and NAT64 Apple app store tests
by Anthony Alba 17 Sep '19

17 Sep '19

Hi folks, I'm wondering what you do to pass Apple app store tests (IPv6-only NAT64/DNS64 scenario) with the iOS app likely to put IPv6 addresses in the SDP, when Kamailio and your B2BUA(Asterisk, FreeSWITCH,...) are IPv4-only endpoints. By "pass" I mean media flowing + SIP working - not sure the app store gatekeepers would actually test media, but it would be nice to have a robust solution. Strictly speaking, this is not a Kamailio's problem but a media SDP/<your-rtp-proxy-of-choice> problem. But since everything is so closely intertwined I'm wondering where you choose to the location the "solution". I have noticed that if Kamailio is IPv4-only and it "sees" IPv6-addresses(Contact, Via headers??) it will try to resolve the IPv6-address as a DNS name! The Wikipedia article on NAT64 mentions 464XLAT(RFC 6877) and Port Control Protocol (RFC 7225), but those seem to be deep rabbit-holes... Cheers Anthony Alba

4 5

Kamailio merchandising shop
by Henning Westerholt 16 Sep '19

16 Sep '19

Hello, there is now a Kamailio merchandising shop available: https://skalatan.de/merchandising/ You find there many options e.g. different Kamailio T-Shirts, Hoodies, Bags, Caps, Mugs, etc.. It provides shipping in many countries world-wide (based from Europe and also USA). Feedback via (direct) e-mail welcome. Best regards, Henning

1 1

3GPP compliance
by Tihomir Culjaga 15 Sep '19

15 Sep '19

Hi, is there any 3GPP compliance list for Kamailio IMS ? Regards, Tihomir.

2 1

RUN kamailio in kubernetes
by Gaurav Bmotra 14 Sep '19

14 Sep '19

hi i want to run kamailio in kubernetes deployment and i want to run this kamailio/kamailio-ci:5.2 image in to the kubernetse .. can anyone help thank you -- *Regards:* Gaurav Kumar

5 6

Problem with incoming audio from Asterisk
by przeqpiciel 14 Sep '19

14 Sep '19

Hi, After a month of fight with SIP signalling I made it but now I have problem with incoming audio from Asterisk. Kamailio and Asterisk are their own public IPs I dont use rtpproxy. When Asterisk sends rtp I'm able to see RTP on my router but it not going to my computer. Maybe somebody have idea why ?

2 1

Using $shv within current call only
by Zhan Bazarov 13 Sep '19

13 Sep '19

Hello, we ran into a trouble with sharing variable into reply route... We are going to match asterisk's uniqueid with rtpengine records like this: route[from_asterisk]{ $var(UniqueId)=$hdr(X-UniqueId); $var(rtp_flags) = $var(rtp_flags) + " label={uid:$var(UniqueId),stream:outgoing} record-call=on"; rtpengine_manage($var(rtp_flags)); } it nicely work for outgoing request, but doesn't for reply.... reply_route[from_pstn]{ rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing"); } and it doesn't work... The one possible way what we found its using $shv instead of $var/$avp... But its unsecure way, cuz it may confus our bussines logic... Do you have any idea how to implement this, how we can assign $shv within current call only? Many thanks!

3 2

Issue with <null> and 0 when comparing vars
by Joel Serrano 13 Sep '19

13 Sep '19

Hello, I'm running into what I believe a weird situation when comparing <null> to 0. I have the following lines: xlog("L_NOTICE", "DEBUG: sel(contact.expires)=$sel(contact.expires) - hdr(Expires)=$hdr(Expires)\n"); if ($hdr(Expires)==0 || $sel(contact.expires)==0) { xlog("L_NOTICE", "[end] removed user location\n"); } else { xlog("L_NOTICE", "[end] saved user location\n"); } I registered with 3 devices, one sends expires as header, another as contact param, and the last uses both header AND contact param. My surprise was this: 1- Both header and contact param: Sep 12 19:37:48 csbc01 csbc[8036]: NOTICE: {1 4 REGISTER 02c3ea6ee623916d37532188031017c9@0:0:0:0:0:0:0:0} <script>: DEBUG: sel(contact.expires)=600 - hdr(Expires)=600 Sep 12 19:37:48 csbc01 csbc[8036]: NOTICE: {1 4 REGISTER 02c3ea6ee623916d37532188031017c9@0:0:0:0:0:0:0:0} <script>: [end] saved user location 2- Only contact param: Sep 12 19:36:15 csbc01 csbc[8035]: NOTICE: {1 40629 REGISTER f2cbaa53-eabfd633(a)84.121.150.224} <script>: DEBUG: sel(contact.expires)=600 - hdr(Expires)=<null> Sep 12 19:36:15 csbc01 csbc[8035]: NOTICE: {1 40629 REGISTER f2cbaa53-eabfd633(a)84.121.150.224} <script>: [end] removed user location 2- Only header: Sep 12 19:29:33 csbc01 csbc[8033]: NOTICE: {1 402 REGISTER 2_1804628355(a)192.168.30.31} <script>: DEBUG: sel(contact.expires)=<null> - hdr(Expires)=3600 Sep 12 19:29:33 csbc01 csbc[8033]: NOTICE: {1 402 REGISTER 2_1804628355(a)192.168.30.31} <script>: [end] removed user location I can workaround the header by using is_present_hf() before comparing, I can also probably parse the $ct to see if it has some Expires in it, but I really want to know if my understanding is incorrect and I'm doing something wrong or if this is a bug? Thanks! Joel.

2 2

Testing server up with SIP OPTIONS/netcat
by Andrew White 13 Sep '19

13 Sep '19

Hi all, As part of a custom monitoring solution, I’m writing a simple script that sends an OPTIONS via netcat to a Kamailio server. However the server doesn’t appear to like it. Here’s the text - options.sip: OPTIONS sip:10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.20:5060 From: <sip:monitoring@10.0.0.20> To: sip:monitoring@10.0.0.10 Call-ID: monitoring(a)10.0.0.20 CSeq: 1 OPTIONS I’m running the following command from the 10.0.0.20 machine: nc -u 10.0.0.10 5060 < options.sip And I get the following error in the Kamailio logs: ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] WARNING: <core> [core/receive.c:319]: receive_msg(): parsing relevant headers failed ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: pv [pv_core.c:1893]: pv_get_hdr(): error parsing headers ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} maxfwd [mf_funcs.c:51]: is_maxfwd_present(): parsing MAX_FORWARD header failed! ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/msg_translator.c:2361]: build_res_buf_from_sip_req(): alas, parse_headers failed As far as I can see, the message itself is valid. It appears parse_cseq() is expecting a higher cseq, or more headers? I’ve looked at another OPTIONS message we get regularly from one of our upstream, and it’s almost identical to this one. Can anyone point me the right way here? Thanks! Andrew

3 2

Presence and BLF
by Bill Neely 12 Sep '19

12 Sep '19

Hi all: I am trying to set up presence and BLF notifies on Kamailo 5.2.4 I am following the instructions laid out in http://kb.asipto.com/kamailio:presence:k43-blf Don't know if these instructions are valid, but I am definitely not getting any light action (using Grandstream 2130 and Grandstream Wav for testing.) Entries in active_watchers appear to be correct. Here is my cfg file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v5.1 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** 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 block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" #!define WITH_DEBUG #!define WITH_NAT #!define WITH_AUTH #!define WITH_MYSQL #!define WITH_USRLOCDB #!define WITH_PRESENCE ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_DLG 9 #!define FLT_DLGINFO 10 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=3 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:62.151.xx.xx:50060 # allow http requests listen=tcp:127.0.0.1:8080 tcp_accept_no_cl=yes #!ifdef WITH_TLS enable_tls=yes #!endif /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "104.192.xx.xx" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # load the module loadmodule "xhttp.so" loadmodule "json.so" loadmodule "pua_json.so" loadmodule "dialog.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:62.151.xx.xx:50060" ) modparam("presence", "send_fast_notify", 0) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 2) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) # ----- presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_single_dialog", 0) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) #!endif # ----- dialog params ----- modparam("dialog", "db_url", DBURL) modparam("dialog", "enable_stats", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { xlog("L_ALERT","ALERT: REQUEST ROUTE ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^6 $rm from $fu (IP:$si:$sp) $du\nMESSAGE BUFFER:\n$mb\n\n\nRU:$ru\n"); $var(i)=0; while($var(i)<$branch(count)) { xlog("$(branch(uri)[$var(i)])\n"); $var(i) = $var(i) + 1; } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle registrations route(REGISTRAR); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); xlog("L_ALERT","ALERT: line 492******************************** $rm from $fu (IP:$si:$sp)\n"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_ALERT","ALERT: line 494******************************** $rm from $fu (IP:$si:$sp)\n"); record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } event_route[xhttp:request] { if ($hu != "/presence/") { xlog("L_INFO", "invalid request [$hu]"); xhttp_reply("404", "Not Found", "text/html", ""); return; } $var(call-id) = $(rb{json.parse,Call-ID}); xlog("L_INFO", "$var(call-id)|log|[$hu] payload $rb"); if ($(rb{json.parse,Event-Package}) == "dialog") { xlog("L_INFO", "$var(call-id)|log|received $(rb{json.parse,Event-Package}) update for $(rb{json.parse,From}) $rb"); pua_json_publish($rb); } xhttp_reply("200", "OK", "text/html", "$rb"); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); dlg_manage(); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!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"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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 route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { xlog("ALERT:NATMANAGE********************* (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("corwie"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains # Routing to foreign domains route[SIPOUT] { xlog("ALERT: SIPOUT (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } route[TOAST] { rewritehostport("45.33.xx.xx:5060"); $fu = "sip:7028743833@62.151.182.83:5012"; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

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