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39 participants
37513 discussions

failure_route, change_reply_status, and append_hf: Not Working as Expected
by sadik.oualla.mohamed＠gmail.com 26 Sep '23

26 Sep '23

Hello Kamailio Community, I'm currently working on a project that involves complex SIP routing and handling. I've encountered some difficulties while using change_reply_status, and append_hf functions in failure_route route in Kamailio. Although the documentation does provide guidance, the behaviors I am experiencing aren't aligning with the expected outcomes. Issue 1: failure_route I've set up a failure_route to handle specific SIP response codes (>4xx). However, I've observed that it get trigered and lunched, but I can't use change_reply_status or append_hf, I was using this first in onreply_route. Issue 2: change_reply_status I tried to use the change_reply_status function to modify the status of SIP replies, but it raises an error that this command cannot be executed here. Issue 3: append_hf I've also attempted to use append_hf in my failure_route block to add headers to replies, but the headers are not being added as expected. As a workaround, I've been using send_reply and append_to_reply for adding custom headers, and they seem to be working as expected. However, I found some old messages in kamailio mailing list, I can summerize what I learned like this ``` #Functions and Context - append_hf(): Adds a header to the currently processed SIP message. If you are in a route {...} block, then this adds the header to the request. If you are in an onreply_route, it adds the header to the reply. This function always acts on the message currently under scrutiny. - append_to_reply(): This function is used to add a header to a reply that will be generated by Kamailio at a later time. This function only affects replies generated by Kamailio itself and is not applicable for messages that are simply being forwarded. #Contexts - route {...}: This is typically where the incoming SIP request gets processed. Here, append_hf() will add headers to that request, and append_to_reply() will add headers to any replies that Kamailio might generate for this request. - onreply_route: This is where incoming SIP replies are processed. Here, append_hf() will act on the reply, not the request. ``` I'd like to know if there is a way to handle sip replies in failure_route, because send_reply make a new reply from kamailio, instead I want to keep some important headers from the original reply. I'd appreciate any insights or guidance the community can offer. Best regards,

2 1

RTPengine use single port for all calls
by Mahmood Alkhalil 26 Sep '23

26 Sep '23

Hello Kamailio, Is it possible to use RTPengine with a single UDP port for all calls instead of a range of ports? For example, when an invite is received or sent, the offered SDP for RTPengine is to receive RTP packets on a single port that is the same for all other calls, so that on firewalls only one port is to be opened.

2 2

Re: Choose RTP port in routing script
by Duarte Rocha 26 Sep '23

26 Sep '23

2 1

cdr_json_pre_encoded_prefix
by Oleg Podguyko 26 Sep '23

26 Sep '23

I try to use cdr_json_pre_encoded_prefix (string) parameters from acc_json module modparam("acc_json", "cdr_extra", "spx_resp=$avp(spx_resp)") modparam("acc_json", "cdr_json_pre_encoded_prefix", "spx_resp") do restart and get next mistakes: sep 22 14:17:15 vms-vms-spx-1 kamailio[581922]: ERROR: <core> [core/modparam.c:169]: set_mod_param_regex(): parameter <cdr_json_pre_encoded_prefix> of type <1:string> not found in module <acc_json> сен 22 14:17:15 vms-vms-spx-1 kamailio[581922]: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio_vmsspx/modules.cfg, line 125, column 63: Can't set module parameter sep 22 14:17:15 vms-vms-spx-1 kamailio[581922]: ERROR: bad config file (1 errors) (parsing code: 0) sep 22 14:17:15 vms-vms-spx-1 systemd[1]: kamailio_vmsspx.service: Control process exited, code=exited status=255 sep 22 14:17:15 vms-vms-spx-1 systemd[1]: kamailio_vmsspx.service: Failed with result 'exit-code'. What’s wrong? I used: [opodguyko@vms-vms-spx-1 ~]$ kamailio -version version: kamailio 5.5.3 (x86_64/linux) b02728-dirty 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 ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: b02728 -dirty compiled on 15:35:46 Dec 15 2021 with gcc 8.3.1 [opodguyko@vms-vms-spx-1 ~]$ This parameter was added at release 5.5.0 Oleg Podguyko

3 3

Choose RTP port in routing script
by Duarte Rocha 26 Sep '23

26 Sep '23

Hello all, I have my rtpengine configured to use ports 20000 - 50000 for RTP in rtpengine.conf. Now i want to able to select the port or the port range in a call by call basis. Is that possible? I've tried to use "media-address" flag on rtpengine_manage but i wasn't successful. Thanks in advance, Cheers, Duarte Rocha

2 1

Several crashs in prepare_new_uac
by Igor Potjevlesch 26 Sep '23

26 Sep '23

Hello! We had several crashs of Kamailio (5.3.4) in the last weeks. Each time, the last logs are: /usr/local/sbin/kamailio[20942]: CRITICAL: <core> [core/forward.c:347]: get_send_socket2(): unsupported proto 0 (*) /usr/local/sbin/kamailio[20966]: CRITICAL: <core> [core/forward.c:347]: get_send_socket2(): unsupported proto 111 (unknown) When we look to the coredump, here below the first traces: (gdb) bt #0 0x00007f70131b3d91 in prepare_new_uac (t=0x7f6fe1d45340, i_req=0x7f7016fbe208, branch=0, uri=0x7fff37142fa0, path=0x7fff37142f80, next_hop=0x7f7016fbe480, fsocket=0x0, snd_flags=..., fproto=0, flags=2, instance=0x7fff37142f70, ruid=0x7fff37142f60, location_ua=0x7fff37142f50) at t_fwd.c:463 #1 0x00007f70131b7c42 in add_uac (t=0x7f6fe1d45340, request=0x7f7016fbe208, uri=0x7f7016fbe480, next_hop=0x7f7016fbe480, path=0x7f7016fbe848, proxy=0x0, fsocket=0x0, snd_flags=..., proto=0, flags=2, instance=0x7f7016fbe858, ruid=0x7f7016fbe870, location_ua=0x7f7016fbe880) at t_fwd.c:805 #2 0x00007f70131bfebd in t_forward_nonack (t=0x7f6fe1d45340, p_msg=0x7f7016fbe208, proxy=0x0, proto=0) at t_fwd.c:1667 #3 0x00007f701316cf44 in t_relay_to (p_msg=0x7f7016fbe208, proxy=0x0, proto=0, replicate=0) at t_funcs.c:332 #4 0x00007f70131a3a11 in _w_t_relay_to (p_msg=0x7f7016fbe208, proxy=0x0, force_proto=0) at tm.c:1689 #5 0x00007f70131a4c51 in w_t_relay (p_msg=0x7f7016fbe208, _foo=0x0, _bar=0x0) at tm.c:1889 #6 0x00000000005a1f57 in do_action (h=0x7fff37143d90, a=0x7f7016bd1b10, msg=0x7f7016fbe208) at core/action.c:1071 #7 0x00000000005aeb1e in run_actions (h=0x7fff37143d90, a=0x7f7016bd1b10, msg=0x7f7016fbe208) at core/action.c:1576 #8 0x00000000005af1df in run_actions_safe (h=0x7fff37146e90, a=0x7f7016bd1b10, msg=0x7f7016fbe208) at core/action.c:1640 #9 0x000000000066aa50 in rval_get_int (h=0x7fff37146e90, msg=0x7f7016fbe208, i=0x7fff37144238, rv=0x7f7016bd1e60, cache=0x0) at core/rvalue.c:915 #10 0x000000000066f000 in rval_expr_eval_int (h=0x7fff37146e90, msg=0x7f7016fbe208, res=0x7fff37144238, rve=0x7f7016bd1e58) at core/rvalue.c:1913 #11 0x000000000066f453 in rval_expr_eval_int (h=0x7fff37146e90, msg=0x7f7016fbe208, res=0x7fff371446ec, rve=0x7f7016bd2588) at core/rvalue.c:1921 #12 0x00000000005a1a1d in do_action (h=0x7fff37146e90, a=0x7f7016bd2e08, msg=0x7f7016fbe208) at core/action.c:1047 #13 0x00000000005aeb1e in run_actions (h=0x7fff37146e90, a=0x7f7016bcbd20, msg=0x7f7016fbe208) at core/action.c:1576 #14 0x000000000059e97d in do_action (h=0x7fff37146e90, a=0x7f7016ef19b8, msg=0x7f7016fbe208) at core/action.c:695 #15 0x00000000005aeb1e in run_actions (h=0x7fff37146e90, a=0x7f7016e3f578, msg=0x7f7016fbe208) at core/action.c:1576 #16 0x000000000059e97d in do_action (h=0x7fff37146e90, a=0x7f7016c220a0, msg=0x7f7016fbe208) at core/action.c:695 #17 0x00000000005aeb1e in run_actions (h=0x7fff37146e90, a=0x7f7016c220a0, msg=0x7f7016fbe208) at core/action.c:1576 #18 0x00000000005aae93 in do_action (h=0x7fff37146e90, a=0x7f7016c23978, msg=0x7f7016fbe208) at core/action.c:1207 #19 0x00000000005aeb1e in run_actions (h=0x7fff37146e90, a=0x7f7016c20748, msg=0x7f7016fbe208) at core/action.c:1576 #20 0x00000000005a1ec6 in do_action (h=0x7fff37146e90, a=0x7f7016c576a0, msg=0x7f7016fbe208) at core/action.c:1062 #21 0x00000000005aeb1e in run_actions (h=0x7fff37146e90, a=0x7f7016c0b770, msg=0x7f7016fbe208) at core/action.c:1576 #22 0x000000000059e97d in do_action (h=0x7fff37146e90, a=0x7f7016bc9920, msg=0x7f7016fbe208) at core/action.c:695 #23 0x00000000005aeb1e in run_actions (h=0x7fff37146e90, a=0x7f7016bb9100, msg=0x7f7016fbe208) at core/action.c:1576 #24 0x00000000005af2a7 in run_top_route (a=0x7f7016bb9100, msg=0x7f7016fbe208, c=0x0) at core/action.c:1661 #25 0x00000000005bcb7a in receive_msg ( buf=0xadbd80 <buf.6971> "INVITE sip:AAAAAAAAAA@W.X.Y.Z SIP/2.0\r\nVia: SIP/2.0/UDP W.X.Y.Z;rport;branch=z9hG4bKZD8Za7p2XF7cF\r\nMax-Forwards: 69\r\nFrom: \"AAAAAAAAAA\" sip:AAAAAAAAAA@W.X.Y.Z;tag=aXy4Na2NtvKBK\r\nTo: <"..., len=1011, rcv_info=0x7fff371474c0) at core/receive.c:423 #26 0x0000000000488e7f in udp_rcv_loop () at core/udp_server.c:548 #27 0x000000000042650f in main_loop () at main.c:1673 #28 0x000000000042ec18 in main (argc=7, argv=0x7fff37147c38) at main.c:2802 Is there anything relevant in this to know what's wrong? Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

3 4

Planning Kamailio Developers Meeting 2023
by Daniel-Constantin Mierla 26 Sep '23

26 Sep '23

Hello, we are considering to organize another edition of Kamailio Developers Meeting, to be hosted again by sipgate.de in Dusseldorf, Germany. There is no fee for participants (i.e., free entry), but each one has to take care of travelling and accommodation (drinks and food are provided during the day). A few developers are already on board, we are looking to see if anyone else from the broader community wants to join us. Likely it is going to be during 2 days (with the option of doing it during three days still open at this moment, if there is going to be enough interest to have it longer). The proposed time frames are: - Nov 7-9, 2023 - Nov 21-23, 2023 The event is about discussing the internals of Kamailio (e.g., architectural changes, development strategies) and writing code or documentation for it. This is not an event about learning how to use Kamailio, but to work on improving it. As a bonus, there will be a social networking event in the evening with some fun activities, drinks and food. If you want to join, state your preference for any of the two time frames (or even just for 2 days in one of the two time frames). It is not a large meeting room, if we get too many join requests, we may have to select based on past contributions, and if that is not the case, then is just the first come first served. You can reply to this thread or just write me directly if you want to join and when you prefer to happen. By mid of next week, we plan to select the event dates so everyone that comes to it can start planning the trip. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy - Training Services -- asipto.com Kamailio World Conference - kamailioworld.com

1 1

Iterating Through Multiple IP Peers for DID Call Termination in Kamailio
by Shah Hussain Khattak 26 Sep '23

26 Sep '23

Hello Team, I hope you're all doing well. I'm reaching out to request your insights on the following scenario. Overview: I have successfully configured Kamailio to authenticate IP addresses, primarily for PBXs and customers unable to register. With this setup, I can flawlessly receive incoming traffic and forward it to our internal FreeSwitch servers. Current issue: Now, I'm focusing on managing traffic in the opposite direction—specifically for DIDs. I have a DID table set up in Kamailio that associates DID numbers with their respective IP peers. Upon receiving a call, a database lookup is performed to find the destination number, after which the call is terminated at the customer's IP. I'm facing difficulty in handling situations where a customer has multiple IP peers (for redundancy). My aim is to try each of these IPs sequentially when terminating a specific DID call towards them. The current logic I'm employing for the INVITE lookup looks like this: # Database lookup for INVITEs if (is_method("INVITE")) { sql_query("didrouting_db", "SELECT route_to FROM did_routing WHERE did_number='$rU'", "result"); if ($dbr(result=>rows) > 0) { $var(route_to) = $dbr(result=>[0,0]); xlog("L_INFO", "Routing DID $rU to $var(route_to)\n"); $du = "sip:" + $var(route_to) + ":5060"; } else { $du = "sip:" + FS_IP + ":5060"; } } The above logic works fine for a single IP. I intend to use a comma-separated list of IPs in my route_to column (for example 192.168.1.2,192.168.1.3, ...) and then iterate through these IPs one by one. I'm having trouble coming up with a workable logic or loop to achieve this. Would you be able to offer any guidance or suggestions? Please also recommend if there is a better approach available to handle this scenario. Thank you in advance for your valuable input. Regards, Shah Hussain

4 3

TM max_inv_lifetime being reset by downstream responses
by James Browne 25 Sep '23

25 Sep '23

Hi all I see this old thread from 2015. - https://kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio.org/t… It's about max_inv_lifetime not working properly. I have this problem today: SIP-180 responses from downstream apparently reset max_inv_lifetime, and the transaction can live for hours (apparently indefinitely) even though max_inv_lifetime is set to just over a minute. By spacing the SIP-180 responses (from downstream) slightly longer apart than than max_inv_lifetime in kamailio, I don't get the problem, so it looks to me that this is TM not matching what's in the documentation (which mentions that this should be a hard limit). With this configuration, SIP-180 responses every minute result in the transaction never ending (until the client CANCELs the call). - kamailio.cfg:modparam("tm", "fr_timer", 3000) - kamailio.cfg:modparam("tm", "fr_inv_timer", 300000) - kamailio.cfg:modparam("tm", "max_inv_lifetime", 65000) - kamailio.cfg:modparam("tm", "noisy_ctimer", 1) That other thread ends with Daniel mentioning that he'd have to look at the code, but nothing further. Running "git log --grep max_inv_lifetime" shows me that the last/only update that wasn't just documentation was in 2007. Has anyone got an ideas? James

2 2

Modifying SDP as drop-in replacement for overloaded Asterisk box - looking for help/paid consulting fast
by Markus 23 Sep '23

23 Sep '23

Hi list, I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted emergency situation (didn't monitor traffic growth properly and now encountering packet loss during peak times) as a drop-in replacement for an overloaded Asterisk box in a call-termination-to-upstream-carrier scenario. My test scenario is to make a call from a SIP softphone to Asterisk IP 1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3 sngrep on Kamailio box 2.2.2.2 - the following SDP will not work - carrier is rejecting it. Carrier is authenticating our calls based on our IP address 2.2.2.2, no username/pass involved. 2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060 INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0 Record-Route: <sip:2.2.2.2;lr> Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8 Max-Forwards: 69 From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd To: <sip:+32xxxxxxxx@2.2.2.2;user=phone> Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060> Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 20.0.0 Date: Fri, 22 Sep 2023 00:06:50 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone> Content-Type: application/sdp Content-Length: 314 X-SIP: 1.1.1.1 v=0 o=root 1093000903 1093000903 IN IP4 1.1.1.1 s=Asterisk PBX 20.0.0 c=IN IP4 2.2.2.2 t=0 0 m=audio 25742 RTP/AVP 8 9 0 101 a=maxptime:150 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:25743 a=ptime:20 I'm comparing this rejected INVITE to a successful INVITE sent by the original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier without Kamailio in the path, and these are the differences I noticed, and probably the things I have to mimick with Kamailio in order to make it work: INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0 should be INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0 To: <sip:+32xxxxxxxx@2.2.2.2;user=phone> should be To: <sip:+32xxxxxxxx@3.3.3.3;user=phone> From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd should be From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060> should be Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060> Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060 should be Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060 o=root 1093000903 1093000903 IN IP4 1.1.1.1 should be o=root 1093000903 1093000903 IN IP4 2.2.2.2 My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU These are the Asterisk boxes I want to originate calls from to Kamailio: [root@voip30 ~]# kamctl address show +-----+-----+----------+------+------+-----------+ | id | grp | ip_addr | mask | port | tag | +-----+-----+----------+------+------+-----------+ | 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv | | 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv | | 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv | | 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv | | 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv | | 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv | | 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv | | 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv | | 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv | +-----+-----+----------+------+------+-----------+ This is the upstream carrier I want Kamailio to proxy calls to: [root@voip30 ~]# kamctl dispatcher show dispatcher gateways +----+-------+------------------+-------+-------+------------+------+ | id | setid | destination | flags | prio. | attrs | desc | +----+-------+------------------+-------+-------+------------+------+ | 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | | +----+-------+------------------+-------+-------+------------+------+ (output manually slightly modified to look properly over E-Mail) As you might have guessed I'm a Kamailio noob... and don't have the resources to learn it as fast as I must to avoid further packet loss. If there's anyone available who can help me to get this done today, optionally in exchange for money, I'd be grateful. Thank you! Markus

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