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37511 discussions

htable write access state
by Patrick Karton 21 Feb '24

21 Feb '24

Hello are htable protected for multiple write access? can i safely update htable from one process and be sure another process will access it for write operation ? if yes is the lock on all the htable or only on the bucket where the key belongs ? Thanks.

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rpm kamailio git master version
by Henning Westerholt 20 Feb '24

20 Feb '24

Hello, just a question about the https://rpm.kamailio.org/ repository, especially regarding git master builds. It seems that the git master build there was updated last autumn. It would be great if it could be updated, now as we want to test for the coming release. Thank you, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

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dialog:event unknown and kemi callback error
by Duarte Rocha 20 Feb '24

20 Feb '24

Hello all, I'm currently developing some kamailio scripts with app_jsdt by migrating a proxy of mine from Kamailio routing script to Js. I've migrated some parts of my request_route and everything seems to be working fine. I am now trying to migrate my event routes to JS and I started with dialog. dialog parameters : modparam("dialog", "event_callback", "dialogEvent") Code : function dialogEvent(event) { KSR.info("Dialog event " + event + " in JS") } After this i try to make a call and in my logs i have : "Dialog event unknown in JS" "Dialog event dialog:start in JS" Is this normal behaviour that I should ignore and only try to process "start,end,failure" or is this a signal of something wrong in my code? I also have this error in my code : "sr_kemi_cbname_lookup_idx(): index 1 is out of range". Does that mean that i have a JS function that calls a missing function in kamailio script or the opposite? How can I identify the root cause? Thanks in advance, Cheers

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Variable specific to branch reply?
by Benoit Panizzon 20 Feb '24

20 Feb '24

Hi List avp stacking is starting to drive me crazy... Somehow I don't understand what is going on in this situation: I have a main branch, and add two more branches with append_branch() I use a on_branch_route trigger: branch_route[BR_TO_CPE] { [Here, some code setting $var(needbh)] if ("yes" == $var(needbh)) { xlog("L_INFO", "$cfg(route): BR_idx: $T_branch_idx BACKHAUL: YES to $nh(u) \n"); add_rr_param(";rtp=yes"); record_route(); $avp(backhaul) = "yes"; $avp(nexthop) = $nh(u); rtpengine_manage($avp(rtp_to_cpe)); } else { xlog("L_INFO", "$cfg(route): BR_idx: $T_branch_idx BACKHAUL: NO to $nh(u) \n"); record_route(); $avp(backhaul) = "yes"; $avp(nexthop) = $nh(u); } } In the Logs, I see Branches Index 0,1,2 with the desired information. So I have 3 values stacked in the avp backhaul and nexthop, right? When I get a reply, I would like to determine if backhauling is required or not. onreply_route[MANAGE_REPLY] { xlog("L_INFO", "MANAGE_REPLY: for BR_idx: $T_branch_idx\n"); while ($(avp(nexthop)[$var(i)]) != $null) { xlog("L_INFO", "MANAGE_REPLY: DEBUG Branch: $T_branch_idx VAR IDX: $var(i) NH: $(avp(nexthop)[$var(i)]) \n"); $var(i) = $var(i) + 1; } } So on EVERY reply, I would expect to get three "DEBUG" lines as I stacked 3 AVP.. Bug I get only 2! Why I am missing one of the stacked AVP? (the last one as it looks like) Or is there any other way to access a arbitrary variable specific to a branch on the reply to that branch? PS: I don't need $avp(nexthop). It's just an easy way to make sure I get the correct avp index for the branch I am interested in. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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mem_join_free flag in kamailio 5.6 deb
by Arsen Semenov 20 Feb '24

20 Feb '24

Hello! I have an issue with kamailio 5.6.5 after some time of flying there are errors in logs: <core> [core/mem/q_malloc.c:297]: qm_find_free(): > qm_find_free(0x7fc79c286010, 536); Free fragment not found! mem_join=1 by default, but it seems that MEM_JOIN_FREE flag is not defined in this build kamailio -I > Print out of kamailio internals > Version: kamailio 5.6.5 (x86_64/linux) > Default config: /etc/kamailio/kamailio.cfg > Default paths to modules: /usr/lib/x86_64-linux-gnu/kamailio/modules > Compile flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, > DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_BLOCKLIST, > HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED > MAX_RECV_BUFFER_SIZE=262144 > MAX_URI_SIZE=1024 > BUF_SIZE=65535 > DEFAULT PKG_SIZE=8MB > DEFAULT SHM_SIZE=64MB > ADAPTIVE_WAIT_LOOPS=1024 > TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select > Source code revision ID: unknown > Compiled with: gcc 10.2.1 > Compiled architecture: x86_64 > Compiled on: > OS Debian 11 bullseye However I noticed the flag exists in 5.7 installed from deb Print out of kamailio internals > Version: kamailio 5.7.4 (x86_64/linux) > Default config: /etc/kamailio/kamailio.cfg > Default paths to modules: /usr/lib/x86_64-linux-gnu/kamailio/modules > Compile flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, > DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, > *MEM_JOIN_FREE*, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED > MAX_RECV_BUFFER_SIZE=262144 > MAX_URI_SIZE=1024 > BUF_SIZE=65535 > DEFAULT PKG_SIZE=8MB > DEFAULT SHM_SIZE=64MB > ADAPTIVE_WAIT_LOOPS=1024 > TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select > Source code revision ID: unknown > Compiled with: gcc 10.2.1 > Compiled architecture: x86_64 > Compiled on: OS Debian 11 bullseye and now, what got me bit confused is that when I build 5.6.5 from source "make printcdefs" gives me this flag, alongside with other flags which I do not see later in "kamailio -v" output. So I am not sure whether this flag was defined during 5.6.5 deb build time or it has been somehow lost? So I need your advice. Regards, Arsen

1 0

How to query, if callid is handled by rtpengine module already?
by Benoît Panizzon 20 Feb '24

20 Feb '24

Hi I have come across this challenge: Initial invite (no to_tag) has is authenticated and has credentials. So a lookup for the username is made to determine if the customer needs backhauling, thus rtpengine needs to be engaged. On a subsequent invite (re-invite to change codec or to refresh session timer) there is a to_tag so it is being routed without sending a challenge and has no auth username. I don't use dialog on the registrar. This is another transaction. So I don't know, if I need to engage rtpengine on that DSP or not. What is the best way to find out, if rtpengine is already handling that callid? if(rtpengine_query_v("j", "$var(dummy)")) { rtpengine_manage(); } basically works, but generates a couple of ugly error messages in the log as rtpengine_query_v does not find the call. Is there a nicer way? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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TCP timeout occurs when the uac_req_send sets $ouri = Received (sip:10.11.40.XX:57646;transport=tls) but The Received connection normal
by riffs＠sina.cn 20 Feb '24

20 Feb '24

The connection registered by the client is normal, and the client can send messages and renew the registration normally, but the push message to the client using uac_req_send() from kamailio fails. Location information after the client is registered: ]#kamcmd ul.lookup location test01 Received: sip:10.11.40.XX:57646;transport=tls ]#netstat -anp | grep 57646 "tcp 0 0 10.11.55.xx:5061 10.11.40.XX:57646 ESTABLISHED 3467/kamailio" kamailio.lua KSR.pv.sets("$uac_req(ouri)", rows.received) In this way, most users can receive messages normally, but occasionally some users cannot receive messages but can renew their registration and send messages normally. Once this is the case for this user, the use of uac_req_send() will continue to fail until the client establishes a new connection. If a problem occurs, the following logs are displayed on the server: 2024-02-20T10:01:25.329595+08:00 10.11.55.XX (72819) ERROR: <core> [core/tcp_main.c:4692]: tcpconn_main_timeout(): connect 10.11.40.XX:57646 failed (timeout) The connection already exists and should be normal, why does "tcpconn_main_timeout(): connect 10.11.40.XX:57646 failed (timeout)" occur.

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Dispatcher ping: "binding to source address X.X.X.X:XXXX failed: Address already in use" for TCP and TLS destinations
by Jardar Leira 17 Feb '24

17 Feb '24

We have recently upgraded from 5.6.4 to 5.7.4. No change in the KAMAILIO configuration. On all upgraded servers we are now getting this message in the logs for every destination when the dispatcher sends out OPTIONS to destinations listed in the dispatcher list: WARNING: <core> [core/tcp_main.c:1301]: find_listening_sock_info(): binding to source address X.X.X.X:YYYY failed: Address already in use [98] The X.X.X.X being the servers IP address and YYYY being the servers listening port for tcp and/or tls. It's not consistent. For some servers it's every time OPTIONS is sent. For others it's just now and then. udp destinations have no issues. There are no other processes occupying these ports. Changing the listening port on the KAMAILIO also results in the new port reported as in use. Dispatcher OPTIONS still seems to work for both tcp and tls. Unsurprisingly using a dynamic, unprivileged port > 1023 and not the server port as the source port, as it shouldn't. The packages are sendt and a reply is registered. Inbound OPTIONS are processed as expected by KAMAILIO on the listening port. Have I missed something transitioning from 5.6 -> 5.7, or is this a bug? Relevant config: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 2) modparam("dispatcher", "ds_ping_interval", 120) modparam("dispatcher", "ds_ping_from", "sip:ping@xxxxxx.xxx") modparam("dispatcher", "ds_ping_latency_stats", 1) OS: Debian Bullseye Kamailio installed from https://deb.kamailio.org/kamailio57 bullseye Best regards, Jardar Leira

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Forward subscriber username via dispatcher
by s.czech＠nowatel.com 16 Feb '24

16 Feb '24

Hi, I'm trying to forward the original username of the peer via Kamailio dispatcher to the Asterisk and I can't get it working. The scenario looks like this: Client Phone (username 1) -> Asterisk Client ---trunk(test-trunk)--> Kamailio Dispatcher/RTPEngine --trunk--> Asterisk Final Client is dialing via Asterisk Client with a simple extension: exten => _X.,1,Set(CALLERID(all)= 587555611 <587555611>) exten => _X.,n,Dial(SIP/${EXTEN}@test-trunk) Asterisk Client is registering to the Kamailio with its own credentials and Kamailio is registering in the Asterisk Final The problem is that I can't find a way to extract the original trunk username (like test-trunk) in the outgoing connection to the Asterisk Final (via Kamailio): Since i'm using the CALLERID on the Asterisk Client all I can see on the Asterisk Final is: INVITE sip:587322591@192.168.20.74 SIP/2.0 Record-Route: <sip:587322591@192.168.20.74;lr;ftag=as52b874a1;nat=yes> Via: SIP/2.0/UDP 192.168.20.74;branch=z9hG4bKc6e3.a77db1f002191ae3ee1a414b63d34a72.0 Via: SIP/2.0/UDP 192.168.21.228:5060;received=192.168.21.228;branch=z9hG4bK705ec94a;rport=5060 Max-Forwards: 69 From: "587555611" <sip:587555611@192.168.21.228>;tag=as52b874a1 To: <sip:587322591@192.168.20.74> Contact: <sip:587555611@192.168.21.228:5060;alias=192.168.21.228~5060~1> Call-ID: 433f9ff84cdb437f0d383797003c232f@192.168.21.228:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.29.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: "587555611" <sip:587555611@192.168.21.228> Content-Type: application/sdp Content-Length: 268 There is no way I can forward the 'test-trunk' in the SIP header. I tried to add append_hf to ROUTE[relay] like append_hf("X-Trunk $au r\n"); but if the trunk is already registered then $au is null or the CALLERID value. How to extract the original username (test-trunk) on the Asterisk Final? Do you have any ideas? Thank you in advance ;)

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Latest point releases seem to break early dialog UPDATEs from B-side when using topos (again)
by George Diamantopoulos 16 Feb '24

16 Feb '24

Hello all, I've noticed that there seems to be a regression with the topos module, more specifically the redis flavour, but I'm assuming the storage backend shouldn't make a difference. I have confirmed this affects both 5.6.5 and 5.7-nightly, so I'm assuming some backported commit is to blame. Kamailio 5.6.4 used previously, to the best of my memory, was not affected. Early dialog UPDATEs sent from the callee seem to be somehow malformed, since the 200 OK the UAS produces goes directly to the UAC despite having executed record_route() for the original INVITE. I can't imagine how this is possible, I thought responses always honour the Via header and record route shouldn't play a role in this case, right? Simply turning topos off restores the intended behaviour, however, so I can't help but think this is somehow related. Here's two call flows demonstrating the behaviour: * Topos on, 200 OK to UPDATE missing: https://pastebin.com/raw/J0zQeM5g * Topos off, 200 OK to UPDATE routed correctly: https://pastebin.com/raw/49yErezb I was wondering if anyone is aware of any commits that might be responsible for this. Additionally, is there an archive of debian packages with previous point releases so that I can confirm this regression with the latest versions? Thank you! Best regards, George

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