- sr-users - kamailio.org

Roadmap to the next major release - Kamailio v5.4.0
by Daniel-Constantin Mierla 05 May '20

05 May '20

Hello, I am going to send a few emails on different topics to inform the broader community on mailing lists about what was discussed during the last online development meeting that happened last Wednesday, so everyone becomes aware and add own opinions and suggestions. First here is about the next major release, which is considered for the mid of summer. To allow the usual 1-1.5 months of testing time frame, the development has to be frozen in the first part of June (more or less one month from now on). If there will be no other proposals in the near future that end up in postponing the release target date, then try to push to master branch (or do pull requests) any new features that you want in Kamailio 5.4.x in the next 4-5 weeks. Once the code is frozen, you would have to wait for another release cycle to get new features in a stable branch. During the online meeting there was no major missing feature reported, if anyone here knows any, describe it and maybe some developer has time to implement it. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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info: in-memory presentity records
by Daniel-Constantin Mierla 05 May '20

05 May '20

Hello, the presentity records (PUBLISH data) stored by the presence module needed always an external storage backend (sql/no-sql database), for subscriptions (active watchers) it was possible to use memory caching only. I recently added support for cache-only module for PUBLISH data as well, more details are presented in the news article: * https://www.kamailio.org/w/2020/05/in-memory-presentity-records/ That could help simplifying network architecture on distributed platforms leveraging cloud infrastructure and using replications mechanisms (e.g., dmq, evapi+pua_json, nsq, ...) between the Kamailio nodes. The impact on the code was quite significant, given that any PUBLISH triggers a series of internal tasks (apply xcap rules, update watchers, ...), therefore I am calling for community testing to be sure old operational modes were not affected as well as testing the new mode to discover eventual issues -- just open an item on bug tracker when a problem is discovered. Special credits to Fred Posner (lod.com) for supporting the development of this feature and to Emmanuel Schmidbauer for testing and troubleshooting in the early phase of development. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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rtpproxy and NAT
by Pavithra M 05 May '20

05 May '20

Hi, kamailio version - 5.3.3 I have configured rtpproxy in pcscf . rtpproxy 17002 1 0 18:40 ? 00:00:00 /usr/bin/rtpproxy -s udp:10.40.10.3 7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 10.45.4.22 -d DBUG LOG_LOCAL0 I have configured rtpproxy in kamailio.cfg as well. modparam("rtpproxy", "rtpproxy_sock", "udp:10.40.10.3:7722") 10.40.10.3 - private IP 10.45.4.22 - public IP I have enabled NAT configuration also. When i make a call from zoiper client , rtp packets are going to some other public ip Could anyone please help me in this issue. Kindly help.

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load_crl takes 90 seconds to complete it execution with crl file size of 4MB !!!
by mahesh b 03 May '20

03 May '20

Hi All, Am using Kamailio 5.1.9 version, In my tls.cfg i have one client and server profile, along with default client and server profile. I have crl enabled for the non default client and server profile , the crl file size is 4 MB in my case. I have 22 child tcp process. With this what i observe is load_crl is taking close to 90 seconds to finish its execution and return. It is very clear from the code, its because of this for loop. procs_no=get_max_procs(); for(i = 0; i < procs_no; i++) { if (SSL_CTX_load_verify_locations(d->ctx[i], d->crl_file.s, 0) != 1) { ERR("%s: Unable to load certificate revocation list '%s'\n", tls_domain_str(d), d->crl_file.s); TLS_ERR("load_crl:"); return -1; } store = SSL_CTX_get_cert_store(d->ctx[i]); X509_STORE_set_flags(store, X509_V_FLAG_CRL_CHECK | X509_V_FLAG_CRL_CHECK_ALL); } Is there a way this can be enhanced or as per the current kamailio design this is a must to do for each and every profile and the child process array list. Please suggest, is it justifiable for load_crl to take 90 seconds , to complete its execution, with one client and server profile and 22 child tcp process and crl file size of 4MB. More info of my setup: ----------------------------- Kamailio 5.1.9 version is running on the below Linux Kernel version : 3.10.0-693.el7.x86_64 Centos version : CentOS Linux release 7.4.1708 (Core) CPU : 2 cores with model name : Intel(R) Core(TM) i7-6700 CPU @ 3.40GHz [root@miv5000 ~]# cat /proc/meminfo MemTotal: 3882076 kB MemFree: 811244 kB MemAvailable: 2320356 kB Openssl verison : OpenSSL 1.0.2k-fips 26 Jan 2017 Regards, Mahesh.B

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Build-variable name for route[name]
by Karsten Horsmann 02 May '20

02 May '20

Hi List, my google skills dont help me. I know there is a kind of build-in variable that you can use for example to return the name of the route you are referring to right now. here some pseudo kamailio-scripting-language: route[FANCY_ROUTE] { xlog("L_INFO", "$??? something happened\n"; } I search for $??? (as placeholder) which should return the route-name "route[FANCY_ROUTE]" or something like that? Take care. -- Cheers *Karsten Horsmann*

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WebRTC control plane: how do you handle trickle ICE?
by Anthony Alba 01 May '20

01 May '20

Hello, For those of you integrating with WebRTC and using a websocket control plane like SIP.js how do you handle trickle ICE? If you use trickle ICE any recommended RTP media servers (FreeSWITCH, Kurento, ....)? SIP.js does some trick like delay XXX ms before sending out the offer assuming the browser has collected all the ICE candidates, or maybe even wait for the IceGatheringDone event: what do you think of such a pattern? I see that rtpengine source code mentions "trickle" in a few places, but the rtpengine module doesn't make it clear how to send a ICE candidate for an existing call. Cheers Anthony Alba

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Kamailio & rtpproxy with more than one external IP interfaces
by Leonid Fainshtein 30 Apr '20

30 Apr '20

Hi, My server has several external IP interfaces and one internal interface. For some calls I would like to handle RTP via Ext1<->Int interfaces and for other calls: via Ext2<->Int. In order to achieve this, I don't define the '-l' rtpproxy parameter, and depending on the call, I set the required IP address as a parameter of rtpproxy_manage(). I didn't succeed to get the desired result. On my testbed, I have the following interfaces: Ext1: 192.168.0.31 Ext2: 192.168.0.32 Int: 10.159.65.1 When I try to create Ext2<->Int (192.168.0.32<-> 10.159.65.1), the packets that accepted on the internal interface are sent out from the Ext1(192.168.0.31) instead of Ext2 (192.168.0.32). It looks like the decision about the source address is made based on the IP routing table information (?). Also, I would like to have an advertised address for at least one external interface. Is it achievable with just one instance of rtpproxy? Or, it is necessary to run two rtpproxies in the bridged mode wit configured advertize addresses? Thank you in advance, Leonid

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Call from Asterisk - kamailio - Teams
by sip user 30 Apr '20

30 Apr '20

Hi... I have integrated Asterisk with Teams using Kamailio like SBC, thanks this list.. I can make call from Teams to Asterisk throught Kamailio, I have not Callerid from Teams but I'm working in this.. The problem that I have is call from Asterisk to Teams.. So, I have configurated a PSTN in Kamailio to my Asterisk. In Kamailio.cfg I have: * 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 = "IP ASERISK" desc "PSTN GW Address" pstn.gw_port = "PORT ASTERISK" desc "PSTN GW Port" In request_route, when I received an INVITE, I have: if (is_method("INVITE|SUBSCRIBE")) { if(src_ip==IP ASTERISK) { record_route_preset("centralita.arcanos.es:5061 ;transport=$ add_rr_param(";r2=on"); } else { route(PSTN); } } So, If I received an INVITE from a src_ip!=IP ASTERISK, I route to PSTN and call to my Asterisk(works, without callerid, but works). I don't know if there are any best way to do that.. If I received an INVITE from a src_ip==IP ASTERISK, I set record_route_preset, but how can I send the call to Teams? What is wrong? What are I doing wrong? Thanks

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Time taken for tls_fix_domains_cfg initialization to complete is 8 seconds
by mahesh b 30 Apr '20

30 Apr '20

Hi All, I have 10 client profiles and 10 server profiles along with default client and server profile in my tls.cfg These are the numbers on time taken for tls_fix_domains_cfg initialization to complete based on the number of client and server profiles i have. Number of profiles Time diff (Sec) No of tcp child processes Start time End time 5 *3.99* 42 16:47:01.836 16:47:05.826 6 4.69 42 16:59:29.335 16:59:34.027 7 *5.347* 42 17:04:30.271 17:04:35.618 8 *6.10* 42 17:18:41.109 17:18:47.204 9 *6.614* 42 17:22:36.869 17:22:43.483 10 *7.379* 42 17:27:46.676 17:27:54.055 Kamailio 5.1.9 version is running on the below Linux Kernel version : 3.10.0-693.el7.x86_64 Centos version : CentOS Linux release 7.4.1708 (Core) CPU : 2 cores with model name : Intel(R) Core(TM) i7-6700 CPU @ 3.40GHz [root@miv5000 ~]# cat /proc/meminfo MemTotal: 3882076 kB MemFree: 811244 kB MemAvailable: 2320356 kB Openssl verison : OpenSSL 1.0.2k-fips 26 Jan 2017 Now my question is : Is it ok for the api to take "8 seconds" to complete initialization in case of having 10 client and 10 server profiles along with default profiles in tls.cfg ? The time units taken for initialization to complete should it be in the order of milliseconds range or in seconds range ? Are there any more parameters related to tls configuration like "crl " which would play a factor in time taken for the initialization to complete. Currently at my application level , timeout is happening as the initialization is taking more time when more number of profiles are enabled. Please share your inputs on whether it is justifiable with my setup cpu/memory, it is ok for tls_fix_domains_cfg to take 8 seconds for the initialization to complete. Regards, Mahesh.B.

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Next online devel meeting - Wed, Apr 29, 2020, 14:00UTC
by Daniel-Constantin Mierla 29 Apr '20

29 Apr '20

Hello, in order to plan the roadmap to next major release, it is time to organize another online devel meeting, so the developers and community members can syncronize and plan what should be done till Kamailio v5.4 will be out. Because freenode.net service is now requiring user registration to join IRC channels on their servers, we are considering some alternatives that would allow guest participation, one of them being using a matrix chat room. More details about it will be posted soon. I created a wiki page for the event at: * https://www.kamailio.org/wiki/devel/irc-meetings/2020a Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, April 29, 2020, at 14:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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NAT and RTP
by Pavithra M 29 Apr '20

29 Apr '20

Hi, I have installed kamailio as a ims server in openstack and enabled WITH_NAT and rtpproxy . I have installed pcscf, icscf, scscf and hss in four different ubuntu instances . server - kamailio 5.3.3. When i make a call using zoiper client from external network , call is becoming a missed call without giving ACK and BYE. Openstack - pcscf - 10.40.10.3 (floating ip - 10.45.4.22) icscf - 10.40.10.12 scscf - 10.40.10.5 hss - 10.40.10.31 zoiper1 - 10.0.2.15 zoiper2 - 10.0.2.15 kindly help me in this regard. when i installed without using NAT and rtpproxy, the call went well. Since currently i am using Opensatck , it is my prerequisite to use zoiper client externally .so for me it is mandate to enable NAT and rtpproxy. Help me please!!!

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change module parameter in routing logic
by Ali Taher 28 Apr '20

28 Apr '20

Hi, Is it possible to change a module parameter inside the routing logic ? For example I want to change sqlcon parameter of sqlops module based on the originating IP. Regards,

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DNS SRV issue
by Melek Oktay 28 Apr '20

28 Apr '20

Hi All, I have a user, who has a SRV records like below, srv:betaproxy.siptelekom.com 1- SRV betaproxy siptelekom 4 hrs 20 0 5060 beta-proxy-dc01.siptelekom.com (X.Y.87.150) 2- SRV betaproxy siptelekom 4 hrs 10 0 5060 beta-proxy-dc02.siptelekom.com (X.Y.87.160) And I want to send the call to him/her, but one of the potential IP address could be failed. Assume that, first record (X.Y.87.150) is failed when I'm trying to send the packet. Later on, timeout occurs and our Kamailio try to send the packet to second one (X.Y.87.160) after period ot time. My question: Next time when I want to send another call to the same user, how Kamailio could use second (X.Y.87.160) SRV instead of trying to first one (X.Y.87.150) ? What is the elegant way to implement this ? Best Regards, Melek

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RTP relay
by hugues 28 Apr '20

28 Apr '20

Hello I'm trying to run Kamailio in front of an asterisk server due to security issues. Both servers have a public ip address. It works fine but the RTP voice traffic goes directly from the VOIP client to the asterisk server without going through the kamailio server. I want to shut down the asterisk server so that it only talks to the kamailio server. UA -> Kamailio public IP addr <-> Asterisk public ip addr How do I force RTP traffic to go through kamailio. Apparently I don't have NAT so I don't need rtpproxy. But you need something that relays RTP traffic? If anyone has an idea I'll take it. Thank you for your help. Hugues. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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DMQ broadcasting crashes kamailio
by SamyGo 28 Apr '20

28 Apr '20

Hi, Is there a way to broadcast KDMQ to the cluster but not expect a reply back !?as far as I've read the source code dmq_bcast_message is exactly like dmq_send_message in a way that it expects a callback to be executed on response i.e expects a reply. So, the situation I'm facing is I'm broadcasting message to cluster and I do not want a reply back. The following two options result in crash & core dump. 1 - If my script doesn't respond back, by use of dmq_handle_message, it marks the destined servers as "inactive" and stops usrloc sync process which isn't desirable. 2 - If I respond back with the dmq_handle_message it crashes the Kamailio which just received this broadcasted message. Here is how its done in script: *broadcasting message to cluster:* dmq_bcast_message("userOnline", "$fu", "text/plain"); *Receiving and handling a broadcast message:* route[DMQ_HANDLE] { if(!(is_method("KDMQ") || $rm == "KDMQ")) return; if(is_method("KDMQ") || $rm == "KDMQ"){ if($rU =~ "userOnline"){ //user came online in cluster, resume transactions if-any suspended $avp(remoteUser) = $rb; } dmq_handle_message(); exit; } } *Related log lines:* Apr 23 21:15:48 kamailio[916]: ALERT: <script>: [da2c1-2f499] ------ DMQ_HANDLE: UserOnline Event Received ------ Apr 23 21:15:48 kamailio[916]: DEBUG: dmq [message.c:53]: ki_dmq_handle_message_rc(): dmq_handle_message [KDMQ sip:userOnline@9.8.7.123:5060] Apr 23 21:15:48 kamailio[916]: DEBUG: dmq [message.c:66]: ki_dmq_handle_message_rc(): dmq_handle_message peer found: userOnline Apr 23 21:15:48 kamailio[916]: DEBUG: <core> [core/receive.c:437]: receive_msg(): request-route executed in: 401461 usec Apr 23 21:15:48 kamailio[935]: DEBUG: dmq [worker.c:87]: worker_loop(): dmq_worker [2 935] lock acquired and crash/segfault.. Core dump: https://pastebin.com/S7ekCPfF Any help or pointers to solve this would be really appreciated. Best Regards, Sammy

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RFC: C standard requirements
by Daniel-Constantin Mierla 27 Apr '20

27 Apr '20

Hello, (cross-posting because it is relevant to know where users need to run Kamailio as well as what developers tend to use for coding, but replies can go to one or the other list). Somehow I tried to stay close to ANSI-C/C90 with C coding style, for the reason of being compatible as much as possible with old compiles and niche/embedded systems. But I noticed that there are couple of C99 extensions in the core, like inline functions, variadic macros (the debugging macros), use of 'long long' or line comments starting with "//" (not so many, but still such comment style is used). Although I haven't searched for occurrences in core, but I remember seeing designated initializers (or initializing a structure by field names). These are part of C99, as listed by: * https://en.wikipedia.org/wiki/C99#Design Although likely GCC had some of them like custom extensions before the standard was out. On the other hand, newer standard C11 made some of those extensions optional: * https://en.wikipedia.org/wiki/C11_(C_standard_revision)#Optional_features So, my email here is to figure out if we should consider C99 as the requirement for adding C code or be more restrictive because there are still cases of usage on very specific systems that do not have such C compiler. We can eventually make a list of what is accepted or not from the recent versions of C standard to stay as much as possible portable across Unix'es. >From develeopers perspective, it would be good to learn if they use (in other projects, etc.) extensions of the new standards that they found useful and simplify their programming work. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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how to use get my domain name in VIA header (INVITE) instead of IP address.
by Nasida Yuriy 27 Apr '20

27 Apr '20

Hi guys, It sounds simple but I can not get domain name in VIA header (INVITE) instead of IP address. It is just FS ---INVITE --> Kamailio --INVITE--> endpoint Kamailio gets INVITE, and adds one more VIA with it's IP address. How to get domain name instead ? Please advice.

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Multi INVITE issue
by Pavithra M 27 Apr '20

27 Apr '20

Hi, I have configured kamailio IMS in vm based images. I have configured pcscf,icscf,scscf,hss and ue in five different vms.I am able to successfully make a call via IMS. But when making a INVITE from alice to bob, it goes via ue -> pcscf pcscf -> scscf scscf -> icscf icscf -> scscf scscf -> pcscf (here multiple INVITE is happening and giving 483 Too Many Hops error and then the call is going to UE successfully) pcscf -> ue Can anybody help me what am i missing . why multi INVITE is happening there alone and giving me too many hops issue. Even after that issue how the call reaches the other UE. Kindly anyone help me in this regard atleast where to check this scenario.

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Dialog module problem.
by Voip support 24 Apr '20

24 Apr '20

Hello, i am using dialog module in-memory to get all dialogs and show in a website. I use for this kamctl dialog show which is run quite often for autorefresh. I noticed that some calls get hung on kamailio while on caller side and callee side calls are disconnected. I captured a wireshark trace and for same caller IP i get many good calls as well some bad calls (not disconnected properly). When comparing the BYE packets i could not notice any difference. However in the "BAD" calls it looks like the BYE from caller get stuck on kamailio and is not sent/forwarded to callee side. I get BYE retransmissions from caller and all BYE requests got stuck on kamailio. After few seconds Callee side sends BYE and that BYE also get stuck. I am wondering how to trace this issue (differently than taking a wireshark trace) as i dont see any difference in the good BYE and the bad BYE. It looks like for any reason from time to time BYE is not forwarded to the other side and when the other side sents BYE its also not forwarded to call initiator. Please give me a clue how to continue on this. I am using dispatcher module to send the calls. Best regards Tom

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nat_uac_test(8)
by Sergiu Pojoga 24 Apr '20

24 Apr '20

When receiving an INVITE over a specific LTE carrier, I'm seeing 'c=IN IP4 192.0.0.4' in SDP, which isn't technically a RFC1918 or RFC6598 IP address and thus nat_uac_test(8) fails. What elegant workaround can be done to catch such specific cases? Thanks.

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Kamailio like SBC with Teams
by sip user 24 Apr '20

24 Apr '20

Hello good morning ... I am new to this list and I was starting to mess with Kamailio, mainly to set it up as SBC against Teams, in this case. But I can't get it to work for me. If I launch a call from the Teams, in the Kamailio I see: 1.- In syslog: Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 52.114.7.24 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 4160, type 3 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 171:1857:1187, 30 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa1f240, 23, 2, 0x7f90f2438f80), fd_no=17 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0xa1f240, 23, -1, 0x0) fd_no=18 called Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:4219]: handle_tcpconn_ev(): tcp: DBG: sending to child, events 1 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:3902]: send2child(): selected tcp worker 1 10(23159) for activity on [tls:SBC_IP:5061], 0x7f90f2438f80 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1507]: handle_io(): received n=8 con=0x7f90f2438f80, fd=9 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSs<default> Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f90f2438f80 n=1468 fd=9 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf= Apr 15 11:11:41 vps793907 kamailio[23122]: #026#003#003 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9c1700, 9, 2, 0x7f90f2438f80), fd_no=1 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7f90f2438f80, FD 9 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1291]: tcp_read_req(): tcp_read_req: EOF Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9c1700, 9, -1, 0x10) fd_no=2 called Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1434]: release_tcpconn(): releasing con 0x7f90f2438f80, state -1, fd=9, id=30 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1435]: release_tcpconn(): extra_data 0x7f90f2432b40 Apr 15 11:11:42 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:3331]: handle_tcp_child(): handle_tcp_child: reader response= 7f90f2438f80, -1 from 1 Apr 15 11:11:42 vps793907 kamailio[23122]: 13(23167) DEBUG: tls [tls_server.c:604]: tls_h_close(): Closing SSL connection 0x7f90f2432b40 2.- With TCPDUMP: 11:13:09.311797 IP SBC_IP .1024 > SBC_IP .eu.sip-tls: Flags [S], seq 261244614, win 64240, options [mss 1440,nop,wscale 8,nop,nop,sackOK], length 0 11:13:09.311898 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [S.], seq 812357247, ack 261244615, win 29200, options [mss 1460,nop,nop,sackOK,nop,wscale 7], length 0 11:13:09.340358 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [.], ack 1, win 2053, length 0 11:13:09.340560 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [P.], seq 1:187, ack 1, win 2053, length 186 11:13:09.340578 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [.], ack 187, win 237, length 0 11:13:09.341361 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [P.], seq 1:1469, ack 187, win 237, length 1468 11:13:09.369606 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [.], ack 1469, win 2053, length 0 11:13:12.451498 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [S], seq 309084204, win 64240, options [mss 1440,nop,wscale 8,nop,nop,sackOK], length 0 11:13:12.451587 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [S.], seq 3275066862, ack 309084205, win 29200, options [mss 1460,nop,nop,sackOK,nop,wscale 7], length 0 11:13:12.707119 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1, win 2053, length 0 11:13:12.707311 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [P.], seq 1:187, ack 1, win 2053, length 186 11:13:12.707345 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [.], ack 187, win 237, length 0 11:13:12.708273 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [P.], seq 1:1469, ack 187, win 237, length 1468 11:13:12.965101 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1469, win 2053, length 0 11:13:12.965156 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [F.], seq 187, ack 1469, win 2053, length 0 11:13:12.965688 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [F.], seq 1469, ack 188, win 237, length 0 11:13:13.220871 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1470, win 2053, length 0 I can't "receive" anything. I have generated the certificates and configured in the tls.cfg of the Kamailio: [server:default] method = TLSv1.2 verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/ssl/cert.key certificate = /etc/letsencrypt/ssl/cert.crt ca_list = /etc/letsencrypt/ssl/ca.crt Within Kamailio itself I have it configured to return a 200 KeepAlive to Teams when it receives an OPTIONS: event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip: SBC_DNS :5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } And I have measured the record_route for this new one: record_route_preset(" SBC_DNS :5061;transport=tls", " SBC_IP :5061"); I have created a dispatcher.list: # setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional) 1:1:sip:sip.pstnhub.microsoft.com;transport=tls 0 3 socket=tls: SBC_IP :5061;ping_from= sip:SBC_DNS 2:1:sip:sip2.pstnhub.microsoft.com;transport=tls 0 2 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS 3:1:sip:sip3.pstnhub.microsoft.com;transport=tls 0 1 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS 4:1:sip:sip-all.pstnhub.microsoft.com;transport=tls 0 1 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS I think that one of the problems is that I do not send the OPTIONS to the Teams well, since it is on their panel, it indicates that the SBC is INACTIVE. I don't know if you could help me straighten this out a bit ... Thank you so much for everything.. a greeting

5 15

Check if Kamailio is going to CANCEL branch
by Ivan Ribakov 24 Apr '20

24 Apr '20

Hi all, Upon receiving a provisional response from callee in a branch route, is there a way of telling if Kamailio is going to automatically CANCEL that branch because a final reply was received from another branch before that? The closest I got is finding “cancel_b_method” parameter of the TM module, but that is not what I’m after. Previously I was recommended to debug via logging onreply_route but I’m not really sure what information may contain hints of intention to send CANCEL. Does anyone have any suggestions/guesses in which direction I should be digging? Thanks, Ivan

1 0

[info] matrix chat room for kamailio
by Daniel-Constantin Mierla 23 Apr '20

23 Apr '20

Thanks to the efforts made by Fred Posner, a matrix.org server has been installed on Kamailio project infrastructure (matrix.kamailio.dev - server sponsored by lod.com), which enables us to create chat rooms as we need for our community of users and developers. It tries to be an alternative to the existing IRC #kamailio channel, given that a while ago freenode.net had to enforce user registration to join the channels on their service (not complaining about it, they did it for good reasons to protect against spammers and D/DoS attacks). The #kamailio channel on freenode.net will still be in use by the people that like it, there is no plan to consider it obsolete. One of the main benefits of running own matrix server is the ability to allow guest access during the devel online meetings, so people that do not want to create an account can still participate by using a web browser. Also, it offers modern encryption and privacy, along with richer collaboration features (e.g., file sharing), many considering it a viable open source alternative to Slack or other private team collaboration services. The first room created is #kamailio:matrix.kamailio.dev : * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev The matrix server is federated, so you can join if you have an account on any other public federated matrix server or one that you can host yourself. The server on matrix.kamailio.dev is not allowing public registrations. If you are a registered developer (have git commit access) and need a matrix account, let us know and we will find solutions. It is the first phase now and based on the interest from the community, its usage can be increased (e.g., creating rooms for development notifications (commits, pull requests, issue); bridge to irc channel; etc...) or the server can be shut down if proves not useful. We will try to do the next online devel meeting (scheduled for Wednesday, April 29, 2020, at 14:00UTC) on the matrix chat room mentioned above. More details at: * https://www.kamailio.org/w/2020/04/matrix-chat-room-for-kamailio/ Stay safe and healthy! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio, TSILO and "local" calls
by German Cancio 23 Apr '20

23 Apr '20

Dear All, does somebody know up to which point TSILO can handle calls from one UA to another UA, when both are registered on the same Kamailio instance? Is a routing through a backend (such as Asterisk) required in that case? thanks and best regards, Germán

2 2

Integrating Push Notifications with iOS
by German Cancio 22 Apr '20

22 Apr '20

Dear All, we are working on the integration of VoIP Push Notifications with iOS devices via Kamailio. From what we observe, the time window between receiving a VoIP APNS notification that wakes up the client app and the client app being sent back to the background by iOS is extremely narrow. It is around 1-2 seconds, with a (seemingly random) variance of say ~0.5s. So 1-2s is the time window available for getting the client app to properly REGISTER and to receive the INVITE back from Kamailio (using e.g. TSILO); otherwise the call is lost. (On Android, the equivalent time window extends to 10s or more.) Have others observed the same on iOS? We are using a client app that still uses the iOS 12 SDK (with Xcode10). Can we expect changes in that regard with iOS13/Xcode13? many thanks and cheers, Germán

3 5

doubt about xflags
by David Escartin 22 Apr '20

22 Apr '20

Dear all one quick question, reading the module corex doc, seems that xflag are message(transaction) flags. But I made a test and seems for some reason the flag is not seeing activated at the onreply_route, when it's activated on the request route. Seemed more like a script flag behaviour. Maybe I'm missing something? thanks a lot and regards david

2 8

Parallel forking - first responder wins
by Ivan Ribakov 21 Apr '20

21 Apr '20

Hi all, What I’m trying to achieve is to have Kamailio fork an INVITE to multiple endpoints in parallel but only maintain the branch that responds first (first to respond with 200 OK I guess). I’ve read the TM module documentation on forking (https://www.kamailio.org/docs/modules/stable/modules/tm.html#tm.serial_fork… <https://www.kamailio.org/docs/modules/stable/modules/tm.html#tm.serial_fork…>) and as far as I understood a combination of “seturi()” + “append_branch()” + “t_relay()” command calls will allow me to send multiple forked INVITEs in parallel. What I couldn't find information about in the documentation (please point me to it in case I missed it) is what controls (if any) do I have over forked requests. Do I need to keep track of the branches myself and cancel others when first succeeds or does Kamailio have some kind of setting for implementing such behaviour? Thanks, Ivan

7 12

NAT pinging after service restart
by Jacob Greene 21 Apr '20

21 Apr '20

Hello everyone! When I restart the Kamailio service, NAT pinging for all existing location records stops and does not start until the UA registers again. I'm using db_mode 2, and all of the registrations will be there, but for some reason Kamailio wont restart the pinging cycle. Is it possible to configure Kamailio to restart NAT pinging in the event of a service restart? Thanks for reading! Jake Greene Voice Engineer

1 1

Join us for SIP Chronicles Live #1, featuring Giovanni Maruzelli
by Maxim Sobolev 20 Apr '20

20 Apr '20

Dear Real-Time Friends and Colleagues! As many of you we have been totally devastated that we will have no chance to see you in the next few months to come. :-/ Some people in the community believe it might be years. I don’t necessarily agree with that opinion myself. Over the course of the last few years our team had a great time extending live coverage for some of those events that have been affected, got some experience and equipment. Instead of just waiting for the virus to clear, we decided to organize a series of bi-weekly live casts with some of the speakers that we have hoped to see at those events presenting their latest developments live and then answering questions from the audience. So without further ado, let me introduce our first guest Giovanni Maruzzelli, who is going to introduce his newest project SaraPhone ( https://github.com/gmaruzz/saraphone). Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask Giovanni a question about his project live: https://youtu.be/mF9elIcVGE8 Or if you miss that opportunity, you can always watch the recording later on Sippy Labs channel on YouTube and email Giovanni your question at < gmaruzz(a)gmail.com>. SaraPhone is a bare bone SIP WebRTC phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) Stay healthy, optimistic and productive! Also share, like and subscribe. See you soon!!! Regards, Max

1 0

INVITE forking - uniform way to identify branches
by Ivan Ribakov 20 Apr '20

20 Apr '20

Hi all, I have a prioritised list of endpoints, where endpoints with the same priority should be sent in parallel: - endpoint_1_uri; p = 1 - endpoint_2_uri; p = 2 - endpoint_3_uri; p = 4 - endpoint_4_uri; p = 5 As some of the lower priority endpoints are not known upfront, I can not load entire list into the contact list and use TM module's “forking based on Q value” to take care of the rest. So my plan is to load all endpoints with the same priority and fork them in parallel. If none succeed - obtain next endpoint(s) and repeat the process. As I go through the above process, I need to update some counters each time delivery to one of the endpoints succeeds or fails. To be able to perform these “housekeeping” tasks I need to be able to clearly identify branches and link them to the destination URI (endpoint URI) of the forked INVITE that started them. Ideally I would like to be able to maintain relationship “branch_id” => “destination URI” where “branch_id” is the identifier that is appended to Via branch parameter. What is the best/most reliable way to achieve this? I tried logging and observing the value of $du - it was available just before sending out a forked INVITE, but in branch_failure and onreply_route it was <null>. I tried logging and observing $T_req($du) - I only expected it to work from onreply_route, but it was also <null>. I tried logging and observing $T_branch_idx - while it was showing meaningful data before sending forked INVITE and during branch_failure route, I couldn’t access it from onreply_route so I couldn’t tell which of the branches current reply is for. Are there any more variables that could help me build “branch_id” => “destination URI” relationship and access information about “branch_id” and/or “destination URI” from branch_failure and onreply_route? Apologies in advance for a lengthy description, I did my best to summarise the situation and what I’m trying to achieve. Regards, Ivan

1 0

Re: [SR-Users] (no subject)
by ahmed moghazy 20 Apr '20

20 Apr '20

Excuse me sir, if i want to make integration between kamailio and asterisk to make kamailio could use land line , i followed steps but i got errors i want help , was asterisk is only way to make kamailio use landline (fxs,fxo) Sent from my iPhone > On Mar 10, 2020, at 9:52 PM, ahmed moghazy <ahmed.m.moghazi(a)gmail.com> wrote: >

2 1

LCR and fallback
by Edward Romanenco 20 Apr '20

20 Apr '20

Hi! I would like to implement LCR and fallback capabilities for my Kamailio installation. I'm currently using the dispatcher module with a simple configuration file. As I understood, there are multiple modules (carrier route, dynamic routing) that can be used to achieve that, what I'm aiming for is - * Country-based (or prefix-based) routing with up to 4 different carriers (could be expanded in the future, but I won't have more than a dozen), while also maintaining a fallback route if the 1st priority is not available. * Each of my carriers have his own requirements for headers and formatting, for instance, for one of our carriers, having a '+' prefix before dialing to a non-US number is mandatory, while on the others - it is not, or another example - one of my carriers doesn't allow having a P-Asserted-Identity header in my invite, while others do. I would love to have the option to manipulate headers based on the route that I have chosen, both on the default and fallback routes. * It would be nice to receive SMTP or SNMP alerts if a fallback does occur. * Easy management, having the ability to load and reload the routing table on a monthly basis, once the pricing list is changed. What would be your suggestion? Edward

2 1

How to install the PHONENUM module?
by Edward Romanenco 20 Apr '20

20 Apr '20

Hi, I'm trying to add the PHONENUM module into my Kamailio installation. For this, I've cloned the main branch of Libphonenumber<https://github.com/google/libphonenumber/tree/master/cpp> and followed the installation rules as they appear in the relevant README file: Building and testing the library -------------------------------- $ cd libphonenumber/cpp $ mkdir build $ cd build $ cmake .. $ make $ ./libphonenumber_test It all went through and the library was created, but when I try to make and install the module itself, I'm getting the following error. Looks like the library wasn't included in the building context, can you please lend me a hand and tell me how do I include it? root@kamaillioegress:/usr/local/src/kamailio-5.1/kamailio/src/modules/phonenum# make Compiling cphonenumber.cpp g++ -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -Wall -Wno-write-strings -Wno-deprecated -Wno-unused-function -Wno-sign-compare -Wno-strict-aliasing -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -Wall -DNAME='"kamailio"' -DVERSION='"5.2.0-dev6"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 5.4.0"' -D__CPU_x86_64 -D__OS_linux -DVERSIONVAL=5002000 -DCFG_DIR='"/usr/local/kamailio-5.1/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DMEM_JOIN_FREE -DF_MALLOC -DQ_MALLOC -DTLSF_MALLOC -DDBG_SR_MEMORY -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_IP_MREQN -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -I/opt/local/include -DKAMAILIO_MOD_INTERFACE -DMOD_NAME='"phonenum"' -c cphonenumber.cpp -o cphonenumber.o cphonenumber.cpp:24:65: fatal error: phonenumbers/geocoding/phonenumber_offline_geocoder.h: No such file or directory compilation terminated. Makefile:22: recipe for target 'cphonenumber.o' failed make: *** [cphonenumber.o] Error 1 Edward

2 1

Free Persian Asterisk Book and COVID-19 Pandemic
by Mojtaba 19 Apr '20

19 Apr '20

In the context of the COVID-19 pandemic, Nasim Telecom decided to publish the first six-captures of Persian Asterisk book that was written by me and Mr Najafi, published in 2017. Also it was introduced by Mr. David Duffett in AstriCon 2017 on Florida. It would be useful for all people which understand Persian language, Also all other people could use it by going code commands on it and examples. For downloading it go the following link: https://www.nasimtelecom.com/en/persian-asterisk-book/ So, be careful and safety at first, Stay at home and read your favourite books. With Best Regards. --Mojtaba Esfandiari.S

3 2

rtpenigne & STUN
by Ahed Saed 17 Apr '20

17 Apr '20

Hi, I'm using kamailio 5.3 & rtpengine ( both kamailio and rtpengine resides on public IP on same machine) am using below logic to handle clients behind NAT if(nat_uac_test("8")){ rtpengine_manage("SIP-source-address replace-origin replace-session-connection "); }else{ rtpengine_manage("trust-address replace-origin replace-session-connection "); } I don't want all the rtp packet to go over the server, i just want to make it use the rtpengine just in case it is needed. but am not sure how this can be done using kamailio, from the client side i'll try to use STUN, but i understand STUN doesn't work all the time ( ~90% of the cases it will work) can anyone help me with this ? another question - if one client works with STUN and the other is behind NAT how this should be handled? Cheers

1 0

TCP/TLS connection (id: 0) for WebSocket could not be found
by Mack Hendricks 17 Apr '20

17 Apr '20

Hey All, I get this message when trying to route request to a WebSocket client: Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: <core> [core/msg_translator.c:1762]: check_boundaries(): no multi-part body Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: <core> [core/msg_translator.c:429]: clen_builder(): content-length: 651 (651) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: WARNING: <core> [core/msg_translator.c:2786]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: <core> [core/msg_translator.c:2002]: build_req_buf_from_sip_req(): could not create Via header Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: tm [t_funcs.c:337]: t_relay_to(): t_forward_nonack returned error -2 (-2) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: tm [t_funcs.c:355]: t_relay_to(): -2 error reply generation delayed Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: exec: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=1176 a=24 n=sl_reply_error Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: No error (2/SL) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: exec: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=1178 a=2 n=exit Here is what my location table looks like. It should try to send the call to sip:98.209.240.245:56291;transport=ws *************************** 2. row *************************** id: 124 ruid: uloc-5e9685e8-5650-92 username: 2000 domain: ws-test.com contact: sips:2000@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss received: sip:98.209.240.245:56291;transport=ws path: NULL expires: 2020-04-15 05:03:17 q: -1.00 callid: 7ef0a550-b2b9-24c6-5c3b-8e0c667f5533 cseq: 50096 last_modified: 2020-04-15 04:59:57 flags: 0 cflags: 524352 user_agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 socket: NULL methods: NULL instance: NULL reg_id: 0 server_id: 0 connection_id: -1 keepalive: 1 partition: 18 The Websocket list looks like this: Server01:~# kamcmd ws.dump { connections: { 8: wss:98.209.240.245:56291 -> wss:134.122.27.49:4443 (state: OPEN, last used 47s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } I’m not sure what I’m doing wrong. Any ideas? Thanks in advance -Mack

2 8

Kamailio in osm
by Pavithra M 17 Apr '20

17 Apr '20

Hi, Do anybody have worked orchestrating kamailio as an IMS server through osm by launching vnfs in openstack . If so kindly reply me as I need some inputs on that.

1 0

Dealing with mobile client state while using Push Notification and TSILO
by Hubert Odziemczyk 17 Apr '20

17 Apr '20

Hello! We are trying to setup a call-flow with TSILO and Push Notifications for solution with a mobile SIP client (Linphone), based on presentation by Federico Cabiddu (http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-…). We have an issue with "managing" state of the client (active or not) when there is a call coming. To be more specific, lookup in location table doesn't always give a proper answer, as contact expiration is different from actual lifetime of the app (which also differs between iOS and Android). One of proposed solutions is setting Expires=1 for REGISTER and then always rely on Push Notifications, but this scenario seems to be unreliable with some clients that are in foreground, as Push Notification doesn't trigger REGISTER again. Question is: did you perceive similar problems? What are your solutions to deal with them? Thank you in advance! Best regards, Hubert

2 1

Announcing SaraPhone, SIP WebRTC Open Source business phone
by Giovanni Maruzzelli 16 Apr '20

16 Apr '20

My fellow VoIPers, I am pleased to announce the early availability of: SaraPhone ------------------ SaraPhone is a bare bone SIP WebRTC voice phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, Kamailip, OpenSIPS, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) You an find it in GitHub ( https://github.com/gmaruzz/saraphone ). Anyone interested can play with it :). Have fun, giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 2

kamailio integration with MS teams.
by Nasida Yuriy 16 Apr '20

16 Apr '20

Hi guys, Do you have some expirience with kamailio integration with MS teams ? I follow the instructon from here https://skalatan.de/en/blog/kamailio-sbc-teams TLS part is configured correctly. I also got OPTIONS pings working between MS teams and kamailio by following of this instruction. That is very good. But... There is issues with outbound calls from MS teams to kamailo. MS side send INVITE, kamailio responds with 180 and 200 OK, but looks like MS ignores us. I know that I probably should use record_route_preset function here to modify Record-Route headers to satisfy MS wishes. But nothing helps. I also tried to modify contact header in 200 OK like it was done for OPTIONS - no luck. Anyone has a working example of kamailio files for MS team ? Or at least sucessful pcap with them. I can be wrong but looks like MS doesn't respect RFC at all. Thanks. Regards, Team lead<https://www.odesk.com/companies/Nasida_~0168bc9fd46660bcca> [https://odesk-prod-portraits.s3.amazonaws.com/Companies:1712760:CompanyLogo…]

4 6

Weird dialplan result
by David Villasmil 15 Apr '20

15 Apr '20

Hello guys, So i have the following dialplan (almost literally, 2 records) +----+------+-----+----------+-----------+-----------+-----------+----------+---------------------------------------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+-----+----------+-----------+-----------+-----------+----------+---------------------------------------+ | 10 | 1 | 100 | 0 | 910000000 | 0 | | | 1.2.3.4:5060 | | 11 | 1 | 100 | 0 | 910000001 | 0 | | | sip.somedomain.net:5060 | And with kamcmd: kamcmd> dialplan.translate 1 s:910000000 { Output: Attributes: 1.2.3.4:5060 } and kamcmd> dialplan.translate 1 s:910600000 error: 500 - No translation So that works... now, when i get an INVITE I do: dp_match("1", "$rU"); xlog("[DIALPLAN] '$rU' was translated to '$var(destination)'\n"); if ( $var(destination)!=0 ) { $ru = "sip:" + $fU + "@" + $var(destination); route(RELAY); } And this prints: <script>: [DIALPLAN] '910600000' was translated to '195.114.216.188:5060' What am I doing wrong? Kind of confused here... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Kamailio propagates 180 and 200 OK OUT OF ORDER
by Luis Rojas G. 15 Apr '20

15 Apr '20

Good day, I am testing the dispatcher module, using Kamailio as stateless proxy. I have a pool of UAC (scripts in SIPP) and a pool of UAS (also scripts in SIPP) for the destinations. Kamailio version is kamailio-5.3.3-4.1.x86_64. Problem I have is, if UAS responds 180 and 200 OK to Invite immediately, sometimes they are propagated out of order. 200 OK before 180, like this : UAS is 172.30.4.195:5061. UAC is 172.30.4.195:5080. Kamailio is 192.168.253.4:5070 Difference between 180 and 200 is just about 50 microseconds. My guess is that both messages are received by different instances of Kamailio, and then because of context switches, even though the 180 is received before, that process ends after the processing of 200. However, I had the idea that in order to avoid these problems the kamailio processes synchronized with each other using a shared memory. I tried using stateful proxy and I obtained the same result. By the way, anyone has any idea about how Kamailio's share memory is implemented? It clearly does not use the typical system calls shmget(), shmat(), because they are not shown by ipcs command. Before posting here I googled, but I couldn't find anything related to this. I can't believe I am the only one who ever had this problem, so I guess I am doing something wrong... Please, any help. I'm really stuck on this. Thanks. -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

6 11

Re: [SR-Users] [EXT] UE client for IMS
by Woscek, Martin W. 15 Apr '20

15 Apr '20

https://code.google.com/archive/p/boghe/downloads Starting with version labeled: Boghe_2.0.113.744.sfx.exe<https://storage.googleapis.com/google-code-archive-downloads/v2/code.google…> works with IPv6. From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Pavithra M Sent: Tuesday, April 14, 2020 4:28 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] [SR-Users] UE client for IMS Hi, Could anyone suggest me what UE client can be used to test the normal ims call flow . Currently I am using zoiper client and facing issues of 403 forbidden - domain not served . So kindly help me in this regard so that if I change my UE client it works . Thanks

1 0

UE client for IMS
by Pavithra M 15 Apr '20

15 Apr '20

Hi, Could anyone suggest me what UE client can be used to test the normal ims call flow . Currently I am using zoiper client and facing issues of *403 forbidden - domain not served *. So kindly help me in this regard so that if I change my UE client it works . Thanks

2 1

kamailio integration with MS teams.
by Yuriy Nasida 14 Apr '20

14 Apr '20

Do you have some expirience with kamailio integration with MS teams ? I follow the instructon from here https://skalatan.de/en/blog/kamailio-sbc-teams TLS part is configured correctly. I also got OPTIONS pings working between MS teams and kamailio by following of this instruction. That is very good. But... There is issues with outbound calls from MS teams to kamailo. MS side send INVITE, kamailio responds with 180 and 200 OK, but looks like MS ignores us. I know that I probably should use record_route_preset function here to modify Record-Route headers to satisfy MS wishes. But nothing helps. I also tried to modify contact header in 200 OK like it was done for OPTIONS - no luck. Anyone has a working example of kamailio files for MS team ? Or at least sucessful pcap with them. I can be wrong but looks like MS doesn't respect RFC at all. Thanks.

1 0

H264 codec licence
by Prakash Ganesan 14 Apr '20

14 Apr '20

Hi, We are planning to use kamilio/RTP engine for our WebRTC calls. I would like to know how the H264 codec licences are handled as i know there is a royalty fee associated with this video codec from the server perspective. As of now there is plan to use H264 pass through and vp8/vp9 transcoded to H264 by the kamilio/RTP engine. -- Cheers Prakash

2 1

Using dispatcher module : Modifying message before dispatching with send_udp()
by Luis Rojas G. 13 Apr '20

13 Apr '20

Hello, I am using the dispatcher module, Kamailio version 5.3.3. The point is, I am using send_udp() to forward the message, because I don't want Kamailio to add headers Via nor Record-Route. I am just dispatching the Invite message, and the rest of communication to be made just between the endpoints, not through Kamailio. However, I need to add before a custom header, using append_hf(). It seems, send_udp() really changes nothing at all in the message, not even something added by the script with append_hf(). It works if I use forward(), but not if I use send_udp(). Is there a way to accomplish this? Best regards, -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

2 2

Lifetime of Invite with no Ack
by Duarte Rocha 13 Apr '20

13 Apr '20

Greetings, How much time will Kamailio keep a transaction active if the ACK to a INVITE is never received by Kamailio? Best Regards,

3 7

Using Stir/Shaken with Kamailio
by Alexandru Tripon 13 Apr '20

13 Apr '20

Hi, After I watched the talk about Stir/Shaken(https://vimeo.com/396940577), I wanted to try the module(secsipid), but I couldn't make it work. I built the Kamailio from source using the following steps: https://kamailio.org/docs/tutorials/devel/kamailio-install-guide-git/ . And then built the secsipid following the instruction found here: https://www.kamailio.org/docs/modules/devel/modules/secsipid.html#idm90 . But when I start the Kamailio with the module loaded I get the following error: ` kamailio: ERROR: <core> [core/sr_module.c:325]: version_control(): no version info in module </usr/local/lib64/kamailio/modules/secsipid.so>: /usr/local/lib64/kamailio/modules/secsipid.so: undefined symbol: module_version ` How can I fix this? Thank you in advance, Alexandru Tripon

2 2

Callee BYE routing issue
by Daniel W. Graham 13 Apr '20

13 Apr '20

Have an issue with BYE routing in a lab setup, any suggestions on what I should be looking at? Topology: UA1 <-> NAT device 1 <-> Proxy1 <-> Registrar/B2BUA <-> Proxy1 <-> NAT device 1 <-> UA2 All UA to UA calls flow through B2BUA. All IP’s are public except for UA1 and UA2. -Both endpoints can call each other. -If the caller hangs up first, BYE is routed properly. -If the callee hangs up first, BYE causes proxy to send 478. On calls where BYE results in 478: if(!isdsturiset()) { ……….…………………….not executed handle_ruri_alias() } return; Example issue call: UA2 to UA1, UA1 hang-up. Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=16 n=if Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=532 a=24 n=isdsturiset Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=2 n=return ……….. Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} tm [ut.h:245]: uri2dst2(): bad_uri: [sip::;transport=] Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} tm [t_fwd.c:1732]: t_forward_nonack(): failure to add branches Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=275 a=24 n=sl_reply_error Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} sl [sl_funcs.c:401]: sl_reply_error(): stateless error reply used: Unresolvable destination (478/SL) Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=277 a=2 n=exit 2020/04/10 00:31:28.625060 NAT_Device_IP:57808 -> Proxy_IP:5060 BYE sip:398@B2BUA_IP:5160 SIP/2.0 Via: SIP/2.0/UDP UA1_IP:52884;branch=z9hG4bK-524287-1---43b5f31829f20952 Max-Forwards: 70 Route: <sip:Proxy_IP;lr=on;ftag=as78671fd9;did=b1b.a27> Contact: <sip:399@UA1_IP:52884> To: "Dan Test" <sip:398@B2BUA_IP:5160>;tag=as78671fd9 From: <sip:399@UA1_IP:52884;rinstance=705e0af5f4be485c;alias=NAT_Device_IP:~57808~1>;tag=64f4b712 Call-ID: 24dc84a4592515245177483c2a657042@B2BUA_IP:5160 CSeq: 2 BYE User-Agent: xxxxxxxxxxxxxxxx Content-Length: 0 ######################################### For comparison, here is a BYE from another test call: UA1 to UA2, UA1 hang-up. This BYE was routed correctly. 2020/04/10 00:44:02.688608 NAT_Device_IP:57808 -> Proxy_IP:5060 BYE sip:398@B2BUA_IP:5160 SIP/2.0 Via: SIP/2.0/UDP UA1_IP:52884;branch=z9hG4bK-524287-1---ce36df02e8a80a48 Max-Forwards: 70 Route: <sip:Proxy_IP;lr=on;ftag=9078f232;did=49e.8c01> Contact: <sip:399@UA1_IP:52884> To: <sip:398@B2BUA_IP>;tag=as5874901f From: <sip:399@B2BUA_IP:5060>;tag=9078f232 Call-ID: 103104NGU1NjkxNGZkYmM4MzA3Y2FkMzY1OGNkZTZmMTMyZDU CSeq: 3 BYE User-Agent: xxxxxxxxxxxxxxxx Authorization: Digest username= xxxxxxxxxxxxxxxx Content-Length: 0 -dan

4 7

Kamailio as IMS (version - 5.3.3) Error : 403 Forbidden Domain not served
by Pavithra M 13 Apr '20

13 Apr '20

Hi David, Kindly help me in this regard. I am not getting like it is because of pval issue or some other issue which causes this issue. I have configured kamailio as an IMS server as VM based . PCSCF - 10.x.x.179 ICSCF - 10.x.x.79 SCSCF - 10.x.x.73 HSS - 10.x.x.75 I am using zoiper client for testing the call flow . The *registration is happening properly* in my both zoiper clients for hss configured users (alice and bob). When i am making an *INVITE , *I am getting *403 Forbidden - Domain not served *error from SCSCF. The Call Flow for invite is as follows . call from alice(a)sip.example.com zoiper1 (UE) -> PCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) PCSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> PCSCF (100 trying) SCSCF -> ICSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) ICSCF -> SCSCF (100 trying) ICSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> ICSCF (100 trying) SCSCF -> ICSCF (*403 Forbidden - Domain not served*) ICSCF -> SCSCF (Request: ACK sip:bob@sip.example.com | ) ICSCF -> SCSCF ( 403 Forbidden - Domain not served ) SCSCF -> PCSCF ( 403 Forbidden - Domain not served ) PCSCF -> UE ( 403 Forbidden - Domain not served ) Below are the alias configuration i have done in pcscf,icscf,scscf nodes *PCSCF.CFG* # IP-Adress for incoming SIP-Traffic, in the following format: # SIP / UDP listen=udp:10.x.x.179:4060 # SIP / TCP listen=tcp:10.x.x.179:4060 # SIP / TCP/TLS listen=tls:10.x.x.179:4060 # IPSEC / UDP #!define IPSEC_LISTEN_ADDR "0.0.0.0" #!define IPSEC_CLIENT_PORT 5062 #!define IPSEC_SERVER_PORT 5063 #alias="sip.example.com:4060" alias=pcscf.sip.example.com:4060 #!define MY_WS_PORT 80 #!define MY_WSS_PORT 443 #!define PCSCF_URL "sip:pcscf.sip.example.com:4060" #!define TCP_PROCESSES 8 #!subst "/NETWORKNAME/sip.example.com/" #!subst "/HOSTNAME/pcscf.sip.example.com/" *ICSCF.CFG* # SIP / UDP listen=udp:10.x.x.79:4070 # SIP / TCP listen=tcp:10.x.x.79:4070 # SIP / TCP/TLS listen=tls:10.x.x.79:4071 alias="icscf.sip.example.com:4070" #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "icscf.sip.example.com" *SCSCF.CFG* # SIP / UDP listen=udp:10.x.x.73:4080 # SIP / TCP listen=tcp:10.x.x.73:4080 # SIP / TCP/TLS listen=tls:10.x.x.73:4081 #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "scscf.sip.example.com" #!define URI "sip:scscf.sip.example.com:4080" alias=scscf.sip.example.com:4080 # ENUM-Server to query: #!define ENUM_SUFFIX "scscf.sip.example.com." Kindly let me know where i am missing . Kindly need help in this issue since the registration is happening successfully and dropping in the INVITE level. Kindly help.

1 0

kamailio error
by Pavithra M 13 Apr '20

13 Apr '20

Hi, This is the lvalue.c file where i am facing the issue. kindly help. if (unlikely(ret<0)){ LM_ERR("assignment failed at pos: (%d,%d-%d,%d)\n", rve->fpos.s_line, rve->fpos.s_col, rve->fpos.e_line, rve->fpos.e_col); } else { if(unlikely(_log_assign_action!=NULL)) _log_assign_action(msg, lv); } rval_destroy(rv); return ret; error: if (rv) rval_destroy(rv); return -1;

1 0

Kamailio - 5.3.3 Error : 403 domain not served
by Pavithra M 13 Apr '20

13 Apr '20

Hi , I have configured kamailio as an IMS server as VM based . PCSCF - 10.x.x.179 ICSCF - 10.x.x.79 SCSCF - 10.x.x.73 HSS - 10.x.x.75 I am using zoiper client for testing the call flow . The *registration is happening properly* in my both zoiper clients for hss configured users (alice and bob). When i am making an *INVITE , *I am getting *403 Forbidden - Domain not served *error from SCSCF. The Call Flow for invite is as follows . call from alice(a)sip.example.com zoiper1 (UE) -> PCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) PCSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> PCSCF (100 trying) SCSCF -> ICSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) ICSCF -> SCSCF (100 trying) ICSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> ICSCF (100 trying) SCSCF -> ICSCF (*403 Forbidden - Domain not served*) ICSCF -> SCSCF (Request: ACK sip:bob@sip.example.com | ) ICSCF -> SCSCF ( 403 Forbidden - Domain not served ) SCSCF -> PCSCF ( 403 Forbidden - Domain not served ) PCSCF -> UE ( 403 Forbidden - Domain not served ) Below are the alias configuration i have done in pcscf,icscf,scscf nodes *PCSCF.CFG* # IP-Adress for incoming SIP-Traffic, in the following format: # SIP / UDP listen=udp:10.x.x.179:4060 # SIP / TCP listen=tcp:10.x.x.179:4060 # SIP / TCP/TLS listen=tls:10.x.x.179:4060 # IPSEC / UDP #!define IPSEC_LISTEN_ADDR "0.0.0.0" #!define IPSEC_CLIENT_PORT 5062 #!define IPSEC_SERVER_PORT 5063 #alias="sip.example.com:4060" alias=pcscf.sip.example.com:4060 #!define MY_WS_PORT 80 #!define MY_WSS_PORT 443 #!define PCSCF_URL "sip:pcscf.sip.example.com:4060" #!define TCP_PROCESSES 8 #!subst "/NETWORKNAME/sip.example.com/" #!subst "/HOSTNAME/pcscf.sip.example.com/" *ICSCF.CFG* # SIP / UDP listen=udp:10.x.x.79:4070 # SIP / TCP listen=tcp:10.x.x.79:4070 # SIP / TCP/TLS listen=tls:10.x.x.79:4071 alias="icscf.sip.example.com:4070" #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "icscf.sip.example.com" *SCSCF.CFG* # SIP / UDP listen=udp:10.x.x.73:4080 # SIP / TCP listen=tcp:10.x.x.73:4080 # SIP / TCP/TLS listen=tls:10.x.x.73:4081 #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "scscf.sip.example.com" #!define URI "sip:scscf.sip.example.com:4080" alias=scscf.sip.example.com:4080 # ENUM-Server to query: #!define ENUM_SUFFIX "scscf.sip.example.com." Kindly let me know where i am missing . Kindly need help in this issue since the registration is happening successfully and dropping in the INVITE level. Kindly help.

1 0

Kamailio 5.3.2 crashing continuously
by harneet singh 13 Apr '20

13 Apr '20

Hi All, As part of our very low cps load at 50 calls per second through Kamailio Version 5.3.2 (Dispatcher Module used with Call Load based Routing - Algorithm 10. Machine is a Centos: 7.7 with kernel version 3.10), we are seeing the kamailio continuously crashing. Please see the data below for 3 types of segfaults seen: Crash-1: -------- Apr 3 19:32:25 FE-A07-34-VM6 kernel: kamailio[10382]: segfault at 7ff300000078 ip 00007ff300000078 sp 00007ffe5f2eafc8 error 14 in libbz2.so.1.0.6[7ff331582000+f000] Apr 3 19:32:25 FE-A07-34-VM6 mysqld: 2020-04-03 19:32:25 6131 [Warning] Aborted connection 6131 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10423]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 13 Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10381]: ALERT: <core> [main.c:767]: handle_sigs(): child process 10382 exited by a signal 11 Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10381]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated =============================== Crash-2: ------- Apr 3 19:36:11 FE-A07-34-VM6 kernel: kamailio[12838]: segfault at 7efe00000078 ip 00007efe00000078 sp 00007ffdc876b868 error 15 in zero (deleted)[7efdc407f000+40000000] Apr 3 19:36:11 FE-A07-34-VM6 mysqld: 2020-04-03 19:36:11 6175 [Warning] Aborted connection 6175 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12874]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 18 Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12826]: ALERT: <core> [main.c:767]: handle_sigs(): child process 12838 exited by a signal 11 Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12826]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated ============================ Crash-3: -------- Apr 3 19:40:53 FE-A07-34-VM6 kernel: kamailio[13542]: segfault at 80 ip 000000000065e193 sp 00007ffdae10c1f0 error 4 in kamailio[400000+476000] Apr 3 19:40:53 FE-A07-34-VM6 mysqld: 2020-04-03 19:40:53 6222 [Warning] Aborted connection 6222 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13516]: ALERT: <core> [main.c:767]: handle_sigs(): child process 13542 exited by a signal 11 Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13558]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 38 Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13516]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated Any pointers for resolution are most welcome as we need to quickly resolve this. Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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IPSec and USIM
by Mohammad Elahi 12 Apr '20

12 Apr '20

Hi All, When my VoLTE Mobile is registered to the Kamailio IMS that support IPSec by AKAv1 algorithm, inside the ESP packet is clear text (not encrypted). Is that because my mistake in configuration or for my USIM card? thanks in advance. Mohammad

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kamailio as SBC for Teams
by Aidar Kamalov 10 Apr '20

10 Apr '20

Hello, I read https://skalatan.de/en/blog/kamailio-sbc-teams and trying to use kamailio for teams. But I faced that MS not responding ACK to my 200OK. Somethin is wrong, but I don't know what. I call from teams to kamailio. Can you help me please This is initial INVITE from Teams: 2020/04/09 22:55:17.346092 52.114.7.24:1024 -> KAMAILIO_IP:5061 INVITE sip:+xxxxxxxxx@MY_KAMAILIO_FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: Aidar Kamalov<sip:+xxxxxxxxx@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=de28a5b5333b451a97312097f91f5564 TO: <sip:xxxxxxxx@MY_KAMAILIO_FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: 3d5ad3cdd01457be94d127fa20478cf1 MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.7.24:5061;branch=z9hG4bKdb903e2c RECORD-ROUTE: <sip:sip-du-a-as.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-a-euno.pstnhub.microsoft.com:443 ;x-i=903eeaac-b19e-43a0-8e75-bb222049922a;x-c=3d5ad3cdd01457be94d127fa20478cf1/d/8/d3d2c2e3f3c14f88bb2fe843b4bc70e4> CONTENT-LENGTH: 2426 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.4.3 i.ASEA.5 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY v=0 o=- 0 0 IN IP4 52.114.252.146 s=session c=IN IP4 52.114.252.146 b=CT:50000 t=0 0 m=audio 52939 RTP/SAVP 104 9 111 18 0 8 103 97 13 118 119 101 a=rtcp:52925 a=ice-ufrag:rKBm a=ice-pwd:BkjUnkfIvYV5pjteSteevL1O a=rtcp-mux a=candidate:4 1 UDP 184547839 52.114.252.146 52939 typ relay raddr 31.13.144.55 rport 57780 a=candidate:4 2 UDP 184547326 52.114.252.146 52925 typ relay raddr 31.13.144.55 rport 57781 a=candidate:1 1 tcp-act 2121006590 100.78.247.179 1024 typ host a=candidate:1 2 tcp-act 2121006590 100.78.247.179 1024 typ host a=candidate:2 1 UDP 2130705919 100.78.247.179 39182 typ host a=candidate:2 2 UDP 2130705406 100.78.247.179 39183 typ host a=candidate:3 1 UDP 1694497791 31.13.144.55 57780 typ srflx raddr 100.78.247.179 rport 39182 a=candidate:3 2 UDP 1694497278 31.13.144.55 57781 typ srflx raddr 100.78.247.179 rport 39183 a=candidate:5 1 tcp-act 1684796926 31.13.144.55 9761 typ srflx raddr 100.78.247.179 rport 41775 a=candidate:5 2 tcp-act 1684796926 31.13.144.55 9761 typ srflx raddr 100.78.247.179 rport 41775 a=candidate:6 1 tcp-pass 174453758 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:6 2 tcp-pass 174453758 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:7 1 tcp-act 174846462 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:7 2 tcp-act 174846462 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=x-candidate-info:4 network-type=WWAN a=x-candidate-info:1 network-type=WWAN a=x-candidate-info:2 network-type=WWAN a=x-candidate-info:3 network-type=WWAN a=x-candidate-info:5 network-type=WWAN a=x-candidate-info:6 network-type=WWAN a=x-candidate-info:7 network-type=WWAN a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31 a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31 a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31|1:1 a=rtpmap:104 SILK/16000 a=fmtp:104 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 SILK/8000 a=fmtp:103 useinbandfec=1; usedtx=0 a=rtpmap:97 RED/8000 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=rtpmap:119 CN/24000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:200 and my 200OK to MS is: 2020/04/09 22:55:19.976427 KAMAILIO_IP:5061 -> 52.114.7.24:1024 SIP/2.0 200 OK Record-Route: <sip:KAMAILIO_FQDN_DOMAIN:5061;transport=tls;ftag=de28a5b5333b451a97312097f91f5564;lr=on> Record-Route: <sip:KAMAILIO_IP:5060;ftag=de28a5b5333b451a97312097f91f5564;lr=on> Via: SIP/2.0/TLS 52.114.7.24:5061;rport=1024;branch=z9hG4bKdb903e2c Record-Route: <sip:ENDPOINT_BEHIND_KAMAILIO_IP;line=sr-QyfKA15Kw1rKw1hKEoPljM0oE1VLX10RQ1VLX10gkY37WJufE1BBjJhVE6E6r14VEJ5Nj6ElE1UNjyrNEJrVjnr.AyudmMV1rg52rJrlAyNBkMVNmSE*> Record-Route: <sip:ENDPOINT_BEHIND_KAMAILIO_IP;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.af1;nat=yes> Record-Route: <sip:KAMAILIO_IP;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.42a1;nat=yes> Record-Route: <sip:KAMAILIO_IP:5061;transport=tls;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.42a1;nat=yes> Record-Route: <sip:sip-du-a-as.pstnhub.microsoft.com:5061;transport=tls;lr> To: <sip:+xxxxxxxxxx@KAMAILIO_FQDN_DOMAIN :5061;user=phone>;tag=3795432917-393759697 From: <sip:+xxxxxxxxxx@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=de28a5b5333b451a97312097f91f5564 Call-ID: 3d5ad3cdd01457be94d127fa20478cf1 CSeq: 1 INVITE Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL Content-Type: application/sdp Accept: application/sdp Content-Length: 282 User-Agent: MY_DOMAIN Contact: <sip:KAMAILIO_IP;line=sr-DHZToOtmMdn7MODy9AnmD10TGLpUrL4noJCE4A03DdQE9LUaMXr*> v=0 o=ETPI-MSX1 0 0 IN IP4 KAMAILIO_IP s=sip call c=IN IP4 KAMAILIO_IP t=0 0 m=audio 12176 RTP/SAVP 8 b=AS:64 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtcp:12177 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:03ZmcR+CxxB7k7C8QvWBgseYq7cH7py7VxqfLM8j a=ptime:20 -- Aydar A. Kamalov

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Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Luis Rojas G. 10 Apr '20

10 Apr '20

Hello, I just realized that I had the dispatcher configured using a hash of Call-ID. That means, after recvfrom there must be an extra processing finding the Call-ID header in message, to calculate a hash and then forward() message. The more the processing, the more cases when 200 could arrive before 180. I just changed it to round robin, and the amount decreased a lot, but it's still there. If I send a burst of 1000 messages, about 5 of them leave out of order every time. Best regards, Luis On 4/9/20 1:48 PM, Luis Rojas G. wrote: > Hello, > > I have a lot of experience developing mutithreaded applications, and I > don't see it so unlikely at all that a process loses cpu just after > recvfrom(). It's just as probable as to lose it just before, or when > writing on a cache or just before of after sendto(). If there are many > messages going through, some of them will fall in this scenario. if I > try sending a burst of 100 messages, I see two or three presenting the > scenario. > > Just forward() with a single process does not give the capacity. I'm > getting almost 1000caps. More than that and start getting errores, > retransmissions, etc. And this is just one way. I need to receive the > call to go back to the network (our application is a B2BUA), so I will > be down to 500caps, with a simple scenario, with no reliable > responses, reinvites, updates, etc. I will end up having as many > standalone kamailio processes as the current servers I do have now. > > I really think the simplest way would be to add a small delay to 200 > OK. Very small, like 10ms, should be enough. Simple and it should > work. As Alex Balashov commented he did for the case with ACK-Re-Invite. > > I have to figure out how to make async_ms_sleep() work in reply_route(). > > Thanks for all the comments and ideas > > Best regards, > > Luis > > > > . On 4/9/20 12:17 PM, Daniel-Constantin Mierla wrote: >> >> >> MICONDA(a)GMAIL.COM appears similar to someone who previously sent you >> email, but may not be that person. Learn why this could be a risk >> <http://aka.ms/LearnAboutSenderIdentification> >> Feedback <http://aka.ms/SafetyTipsFeedback> >> >> Hello, >> >> then the overtaking is in between reading from the socket and getting >> to parsing the call-id value -- the cpu is lost by first reader after >> recvfrom() and the second process get enough cpu time to go ahead >> further. I haven't encountered this case, but as I said previously, >> it is very unlikely, but still possible. I added the route_locks_size >> because in the past I had cases when processing of some messages took >> longer executing config (e.g., due to authentication, accounting, ..) >> and I needed to be sure they are processed in the order they enter >> config execution. >> >> Then the option is to see if a single process with stateless sending >> out (using forward()) gives the capacity, if you don't do any other >> complex processing. Or if you do more complex processing, use a >> dispatcher process with forwarding to local host or in a similar >> manner try to use mqueue+rtimer for dispatching using shared memory >> queues. >> >> Of course, it is open source and there is also the C coding way, to >> add a synchronizing mechanism to protect against parallel execution >> of the code from recvfrom() till call-id lock is acquired. >> >> Cheers, >> Daniel >> -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

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t_relay failure when dropping all branches in same destination set
by Sergiu Pojoga 09 Apr '20

09 Apr '20

Hi there, Can someone help figure out why there's this t_relay error, if I decide to drop the single master branch either all branches belonging to the same destination set, even though there are more serial sets to try later? *Error:* *tm [t_funcs.c:337]: t_relay_to(): t_forward_nonack returned error -6 (-6)* *tm [t_funcs.c:355]: t_relay_to(): -6 error reply generation delayed * *Example:* request_route { seturi("sip:a@mydomain.net"); append_branch("sip:b@mydomain.net", "0.5"); append_branch("sip:c@mydomain.net", "0.5"); append_branch("sip:d@10.22.0.30", "1.0"); # append_branch("sip:e@mydomain.net", "1.0"); # no error if I add this branch with same Q=1.0 t_load_contacts(); t_next_contacts(); t_on_branch("check_branch"); t_on_failure("serial"); t_relay(); break; } branch_route[check_branch] { if($rd=="10.22.0.30") drop; } failure_route["serial"] { if (!t_next_contacts()) exit; } t_on_branch("check_branch"); t_on_failure("serial"); t_relay(); } Thanks in advance. --Sergiu

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Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Alex Balashov 09 Apr '20

09 Apr '20

On Thu, Apr 09, 2020 at 02:26:45PM +0100, David Villasmil wrote: > How about just not forwarding the 180 if it’s coming right after the > 200ok? I know it’s a hack, but a 180 is a provisional response > indicating the ring status, not the actual audio for the ring... so if > a 200 is received, the call is already established, no need to forward > the 180. There is a general principle, particularly in PSTN interworking, that signalling should be reliably conserved end-to-end. I think dropping messages is the most cavalier option available, perhaps worth considering only if every other logical possibility has been exhausted first. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Daniel-Constantin Mierla 09 Apr '20

09 Apr '20

On 09.04.20 09:47, Daniel-Constantin Mierla wrote: > [...] >> >> Any idea why Async is not allowed in reply_route()? >> Didn't notice this question in the first place to respond in my previous email. Probably the developer that did it needed it for SIP requests. However, you can make a patch (pull request) to make it work for other cases. The developers will review the changes and if they don't break anything, it will be merged. Cheers, Daniel

2 1

kamailio - IPSEC
by Pavithra M 09 Apr '20

09 Apr '20

Hi, Is there any possibility to disable ipsec_pcscf in kamailio 5.3.3.

2 2

Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Daniel-Constantin Mierla 09 Apr '20

09 Apr '20

Hello, it was a reply to my email where I mentioned the route_locks_size parameter. As he said he looked at that parameter, I assumed it was about the route_locks_size, because there was not other parameter listed in the emails. So using the route_locks_size parameter doesn't require to use dialog module. Cheers, Daniel On 09.04.20 10:29, Henning Westerholt wrote: > > Hello, > > > > I mentioned in some of earlier e-mails as one possible option to track > the state of a dialog and to act depending on it. > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com <https://gilawa.com/> > > > > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *Sent:* Thursday, April 9, 2020 9:48 AM > *To:* luis.rojas(a)sixbell.com; Kamailio (SER) - Users Mailing List > <sr-users(a)lists.kamailio.org>; Henning Westerholt <hw(a)skalatan.de> > *Subject:* Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER > > > > Hello, > > On 08.04.20 23:03, Luis Rojas G. wrote: > > Hello, Daniel, > > > > I looked into that parameter, but I need to use with the dialog > module, and I'm pretty afraid to use that. > > who said or where is written than you need to load the dialog module? > You definitely don't. > > Cheers, > Daniel > > > > I was looking more into the stateless proxy, because I need to > process a lot of traffic. > > > > My target is 4200CAPS. with duration between 90s and 210. Let's > say, 150 seconds. That would mean 630.000 simultaneous dialogs. I > don't think the solution can go that way. > > it would really help me to be able to use completely stateless > proxy plus Async in reply_route(), to introduce an artificial > delay before forwarding 200 OK to Invite.. As someone mentioned, > it would help me on request_route(), for race conditions between > ACK and Re-Invite. > > Any idea why Async is not allowed in reply_route()? > > Best regards, > > > > Luis > > > > On 4/8/20 1:07 PM, Daniel-Constantin Mierla wrote: > > Hello, > > you have to keep in mind that Kamailio is a SIP packet router, > not a telephony engine. If 180 and 200 replies are part of a > call is not something that Kamailio recognize at its core. Its > main goal is to route out as fast as possible what is > received, by executing the configuration file script. Now, a > matter of your configuration file, processing of some SIP > messages can take longer than processing other. And the > processing is done in parallel, a matter of children parameter > (and tcp_children, sctp_children). > > With that in mind, a way to try to cope better with the issue > you face is to set route_locks_size parameter, see: > > * > https://www.kamailio.org/wiki/cookbooks/devel/core#route_locks_size > <https://nam02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.kamai…> > > Probably is what you look for. > > But if you want more tight constraints, like when receiving a > 180 after a 200ok and not route it out, you have to make the > logic in configuration file by combining modules such as > dialog or htable (as already suggested). > > Cheers, > Daniel > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

pcscf logs - error in kamailio
by Pavithra M 09 Apr '20

09 Apr '20

Hi , The error i am facing in kamailio pcscf Apr 9 13:44:31 tel-VirtualBox kamailio[2847]: INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header present. Apr 9 13:44:31 tel-VirtualBox kamailio[2811]: ERROR: <core> [core/lvalue.c:353]: lval_pvar_assign(): setting pvar failed Apr 9 13:44:31 tel-VirtualBox kamailio[2811]: ERROR: <core> [core/lvalue.c:404]: lval_assign(): assignment failed at pos: (106,30-106,48) Apr 9 13:44:31 tel-VirtualBox kamailio[2811]: ERROR: <core> [core/lvalue.c:353]: lval_pvar_assign(): setting pvar failed Apr 9 13:44:31 tel-VirtualBox kamailio[2811]: ERROR: <core> [core/lvalue.c:404]: lval_assign(): assignment failed at pos: (114,31-114,52) kindly help me in this issue .. because of this , it is throwing 403 domain not server error in scscf

1 0

Error in kamailio 5.3.3
by Pavithra M 08 Apr '20

08 Apr '20

Hi, I have configured kamailio as Ims server with pcscf,icscf,scsf and hss in 4 different vms . After that I use zoiper client for testing the call flow . Registration is successful .It goes via UE -> Pcscf -> icscf -> scscf and returns back with 200 OK . Registration is happening with authentication from hss . When I make an INVITE , it goes from UE -> PCSCF -> SCSCF -> ICSCF -> SCSF and s-cscf gives *403 - Domain not served *to icscf and i-cscf returns to p-cscf with the same error . I am not seeing any errors in scscf and i-cscf . In pcscf , I am seeing an error like *Error : <core> [core/lvalue.c] :lval_pvar_assign() : setting pvar failed * *Error : <core> [core/lvalue.c] :* *lval_assign():assignment failed at pos* What should I do? Kindly advise. I am struggling with this issue for a week.kindly do the needful. Version - kamailio - 5.3.3 I need a normal call flow to be established between two clients.kindly help .

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Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Alex Balashov 08 Apr '20

08 Apr '20

On Wed, Apr 08, 2020 at 10:51:25PM +0200, Daniel-Constantin Mierla wrote: > - corex module offers functions to set source address and received > socket for more flexibility why processing on 127.0.0.1 Interesting. Thanks for the tip! -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Alex Balashov 08 Apr '20

08 Apr '20

On Wed, Apr 08, 2020 at 04:54:48PM +0100, David Villasmil wrote: > I have a question on this, since 180 arrived before the 200, shouldn’t it > go out before? I.e.: FIFO? Or is this not happening because there are > multiple processes and one may take the 180 and the other the 200 and they > are processes independently? > > I guess setting the childs to only 1 should prove this theory. In theory, it should all be FIFO, of course. In practice, there are stochastic aspects to OS process scheduling and I/O scheduling which make the ingress and egress of messages arriving at Kamailio simultaneously or substantially simultaneously somewhat indeterminate. And, of course, it depends on how complicated the processing of the messages respectively is, either in computational terms, or external I/O wait, or both. To Steve Davies' point, that's just the nature of the multiprocess architecture, together with the lack of a distributor thread. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Alex Balashov 08 Apr '20

08 Apr '20

On Wed, Apr 08, 2020 at 01:30:21PM +0000, Henning Westerholt wrote: > There is also the option to do an asynchronous sleep (with the async) > module on the message that you want to delay but still processing > other messages during it. We have a buggy endpoint that consistently sends reinvites immediately following the e2e ACK, and substantially simultaneously with the ACK. It quite often happens that the re-invite goes out the door before the ACK, which is of course rejected by the target due to inconsistent state. We solved this a somewhat nasty hack using just the very technique you mentioned -- we use the async features of Kamailio to 'delay' all re-invites about 50 ms. It does solve this problem, if a bit suboptimally and at the cost of eccentricity and complexity. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

KEMI - "record_route_advertised_address" - function export or alternative
by Jacob Greene 08 Apr '20

08 Apr '20

Hello all! I'm wondering if there are any plans to add record_route_advertised_address() function to KEMI? I'm unable to completely finish my migration to KEMI without this function due to server side NAT in our VPC and the record_route header getting automatically set to the advertised address. Is there a workaround I could use to change the record route header? Is anyone aware of any resources we could utilize to help get this fixed? Thanks for reading! Jake, Voice Engineer

2 1

Kamailio - pcscf error
by Pavithra M 07 Apr '20

07 Apr '20

Hi, I have configured kamailio as Ims server with pcscf,icscf,scsf and hss in 4 different vms . After that I use zoiper client for testing the call flow . Registration is successful .It goes via UE -> Pcscf -> icscf -> scscf and returns back with 200 OK . Registration is happening with authentication from hss . When I make an INVITE , it goes from UE -> PCSCF -> SCSCF -> ICSCF -> SCSF and s-cscf gives *403 - Domain not served *to icscf and i-cscf returns to p-cscf with the same error . I am not seeing any errors in scscf and i-cscf . In pcscf , I am seeing an error like *Error : <core> [core/lvalue.c] :lval_pvar_assign() : setting pvar failed * *Error : <core> [core/lvalue.c] :* *lval_assign():assignment failed at pos* What should I do? Kindly advise. I am struggling with this issue for a week.kindly do the needful. Version - kamailio - 5.3.3 I need a normal call flow to be established between two clients.kindly help .

1 0

[kamailio][webrtc issue]
by Bilal Abbasi 07 Apr '20

07 Apr '20

Hi Users, I am new to kamailio, and trying to implement webrtc on kamailio, i am getting following error when i try to login from tryjssip page. Mar 29 21:11:15 kamailio kamailio[20570]: ERROR: <core> [core/tcp_read.c:1535]: tcp_read_req(): bad request, state=7, error=4 buf:#012GET / HTTP/1.1#015#012Host: kamailio..com:7443#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012User-Agent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_15_2) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/80.0.3987.149 Safari/537.36#015#012Upgrade: websocket#015#012Origin: https://tryit.jssip.net#015#012Sec-WebSocket-Version: 13#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: en-US,en;q=0.9,ur;q=0.8#015#012Sec-WebSocket-Key: 88fVCzdsvQPCrwV7gMrWaA==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012#012parsed:#012GET / HTTP/1.1#015#012Host: kamailio..com:7443#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012User-Agent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_15_2) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/80.0.3987.149 Safari/537.36#015#012Upgrade: websocket#015#012Origin: https://tryit.jssip.net#015#012Sec-WebSocket-Version: 13#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: en-US,en;q=0.9,ur;q=0.8#015#012Sec-WebSocket-Key: 88fVCzdsvQPCrwV7gMrWaA==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip I can see following error on chrome console. WebSocket connection to 'wss://kamailio..com:7443/' failed: Connection closed before receiving a handshake response I tried to set tcp_accept_no_cl to value 'no' and this did not helped me either. Can someone please help me with the issue Regards Bilal Abbasi

3 7

Reloading Contacts/Attributes at runtime
by Sergiu Pojoga 07 Apr '20

07 Apr '20

Hi there, Is there some mechanism to reload Contacts *and *their attributes at runtime, without a full Kamailio restart? Known to me tools: *rpc ul.add* will add a new Contact, but not its attributes *rpc ul.flush* will flush from memory to DB, not the other way around Having a combination of registered/permanent Contacts, I can add permanent ones with *ul.add*, however I couldn't find a way to set its attributes other than in DB location, which implies the need for a reload. Thanks in advance. --Sergiu

1 0

Substitution of Contact Header
by Mack Hendricks 07 Apr '20

07 Apr '20

Hey All, I always have issues with substituting out header values. I’ve tried subst and subst_hf. Here is the value that I want to change out: Contact: <sip:52.41.52.34:5060;line=sr-h06Cy7RYRFkYacM0a7epa7sVa7sVRcNCW7IbaGZ9RGlC <sip:52.41.52.34:5060;line=sr-h06Cy7RYRFkYacM0a7epa7sVa7sVRcNCW7IbaGZ9RGlC>> And this is what I want it to be Contact: <sip:1313717555@jump.dsiprouter.net <sip:1313717555@jump.dsiprouter.net>;line=sr-h06Cy7RYRFkYacM0a7epa7sVa7sVRcNCW7IbaGZ9RGlC> I’m using this subst('/^Contact: <sip:(.*);(.*)>$/Contact: <sip:1313717555@jump.dsiprouter.net;\2 <sip:1313717555@jump.dsiprouter.net;\2>>\r/ig’); And it doesn’t work, but putting into a online regular expression tester it looks good as shown below. Any ideas?

3 4

kamailio-pcscf-error
by Pavithra M 07 Apr '20

07 Apr '20

Hi I am facing the error in pcscf and i have attached the error log and tcp dump of pcscf. kindly do the needful.

1 0

Substitution of Contact Header
by Mack Hendricks 07 Apr '20

07 Apr '20

Hey All, I always have issues with substituting out header values. I’ve tried subst and subst_hf. Here is the value that I want to change out: Contact: <sip:52.41.52.34:5060;line=sr-h06Cy7RYRFkYacM0a7epa7sVa7sVRcNCW7IbaGZ9RGlC <sip:52.41.52.34:5060;line=sr-h06Cy7RYRFkYacM0a7epa7sVa7sVRcNCW7IbaGZ9RGlC>> And this is what I want it to be Contact: <sip:1313717555@jump.dsiprouter.net <sip:1313717555@jump.dsiprouter.net>;line=sr-h06Cy7RYRFkYacM0a7epa7sVa7sVRcNCW7IbaGZ9RGlC> I’m using this subst('/^Contact: <sip:(.*);(.*)>$/Contact: <sip:1313717555@jump.dsiprouter.net;\2 <sip:1313717555@jump.dsiprouter.net;\2>>\r/ig’); And it doesn’t work, but putting into a online regular expression tester it looks good as shown below. Any ideas?

2 1

Scscf error
by Pavithra M 06 Apr '20

06 Apr '20

Hi, I have configured p-cscf ,icscf, scscf , hss in different vms and I am able to get the registration successful. I am using zoiper client and kamailio version 5.3.3. After the registration , when I try to make a call, it is showing 403 , Domain not served In my scscf TCP dump . What is this error and how to solve this . Kindly help me asap since I am facing this issue for a week and struggling to resolve it . Thanks.

1 0

Pcscf error
by Pavithra M 06 Apr '20

06 Apr '20

Hi, I have configured p-cscf ,I cscf, scscf ,hss in four different vms. I am getting below errors in my pcscf log . Kindly provide me the solution asap . Kamailio version - 5.3.3 Apr 6 18:02:29 tel-VirtualBox kamailio: INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt as the io watch method (auto detected) Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: rr [rr_mod.c:177]: mod_init(): outbound module not available Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: auth [auth_mod.c:345]: mod_init(): qop set, but nonce-count (nc_enabled) support disabled Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: path [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: path [path_mod.c:151]: mod_init(): outbound module not available Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:264]: mod_init(): Successfully bound to PCSCF Usrloc module Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:271]: mod_init(): Successfully bound to TM module Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:114]: ipsec_print_all_socket_lists(): Listening on: Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:164]: ipsec_print_all_socket_lists(): udp: 10.45.6.179 [10.45.6.179]:4060 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:164]: ipsec_print_all_socket_lists(): udp: 0.0.0.0 [0.0.0.0]:5062 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:164]: ipsec_print_all_socket_lists(): udp: 0.0.0.0 [0.0.0.0]:5063 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:164]: ipsec_print_all_socket_lists(): udp: 0.0.0.0 [0.0.0.0]:5064 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:164]: ipsec_print_all_socket_lists(): tcp: 10.45.6.179 [10.45.6.179]:4060 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:164]: ipsec_print_all_socket_lists(): tcp: 0.0.0.0 [0.0.0.0]:5062 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:164]: ipsec_print_all_socket_lists(): tcp: 0.0.0.0 [0.0.0.0]:5063 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:164]: ipsec_print_all_socket_lists(): tcp: 0.0.0.0 [0.0.0.0]:5064 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:275]: mod_init(): Successfully bound to PCSCF IPSEC module Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: ERROR: rtpengine [rtpengine.c:2901]: select_rtpp_set(): no rtpp_set_list->rset_first Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: NOTICE: rtpengine [rtpengine.c:1722]: mod_init(): Default rtpp set 0 NOT found Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [main.c:2779]: main(): processes (at least): 82 - shm size: 67108864 - pkg size: 8388608 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: WARNING: tm [tm.c:507]: fixup_routes(): t_on_failure("NATMANAGE"): empty/non existing route Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Apr 6 18:02:29 tel-VirtualBox kamailio[9603]: ERROR: <script>: event_route[htable:mod-init] { Apr 6 18:02:29 tel-VirtualBox kamailio[9722]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Apr 6 18:02:29 tel-VirtualBox kamailio[9730]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:02:29 tel-VirtualBox kamailio[9737]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:02:29 tel-VirtualBox kamailio[9736]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:02:29 tel-VirtualBox kamailio[9738]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:02:29 tel-VirtualBox kamailio[9739]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:02:29 tel-VirtualBox kamailio[9740]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:02:29 tel-VirtualBox kamailio[9735]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/9735 Apr 6 18:02:29 tel-VirtualBox kamailio[9741]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:02:29 tel-VirtualBox kamailio[9752]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:02:29 tel-VirtualBox kamailio[9751]: WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=843 a=5 n=route Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=902 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=902 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=904 a=27 n=sql_query Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=905 a=25 n=xlog Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: <script>: Preloading NAT-PING. Rows: 10 Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=921 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=907 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=919 a=23 n=while Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=911 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=915 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=917 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=918 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=921 a=25 n=sql_result_free Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=845 a=26 n=sht_iterator_start Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=855 a=23 n=while Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=846 a=25 n=sht_iterator_next Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=847 a=25 n=xlog Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: <script>: OPTIONS to sip:bob@10.45.6.75:60060;rinstance=a59e664cda799366;transport=tcp via sip:10.45.6.75:38891;transport=tcp... Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=848 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=849 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=850 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=851 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=852 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=853 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=854 a=24 n=uac_req_send Apr 6 18:03:29 tel-VirtualBox kamailio[9753]: ERROR: <core> [core/tcp_main.c:4456]: handle_tcpconn_ev(): connect 10.45.6.75:38891 failed Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=846 a=25 n=sht_iterator_next Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=847 a=25 n=xlog Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: <script>: OPTIONS to sip:alice@10.45.6.174:43244;rinstance=72779c915b29d0c2;transport=UDP via sip:10.45.6.174:43244... Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=848 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=849 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=850 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=851 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=852 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=853 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=854 a=24 n=uac_req_send Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=846 a=25 n=sht_iterator_next Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=847 a=25 n=xlog Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: <script>: OPTIONS to sip:bob@10.45.6.75:46205;transport=TCP;rinstance=f2285c01dd4acb41 via sip:10.45.6.75:46205;transport=tcp... Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=848 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=849 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=850 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=851 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=852 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=853 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=854 a=24 n=uac_req_send Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=846 a=25 n=sht_iterator_next Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=847 a=25 n=xlog Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: <script>: OPTIONS to sip:bob@10.45.6.75:37861;transport=TCP;rinstance=e965f9424c56508e via sip:10.45.6.75:37861;transport=tcp... Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=848 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=849 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=850 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=851 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=852 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=853 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=854 a=24 n=uac_req_send Apr 6 18:03:29 tel-VirtualBox kamailio[9605]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=861 a=25 n=xlog Apr 6 18:03:29 tel-VirtualBox kamailio[9605]: ERROR: <script>: request sent to sip:alice@10.45.6.174:43244;rinstance=72779c915b29d0c2;transport=UDP completed with code: 200, Type 1 Apr 6 18:03:29 tel-VirtualBox kamailio[9605]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=888 a=16 n=if Apr 6 18:03:29 tel-VirtualBox kamailio[9605]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=887 a=63 n=assign Apr 6 18:03:29 tel-VirtualBox kamailio[9753]: ERROR: <core> [core/tcp_main.c:4456]: handle_tcpconn_ev(): connect 10.45.6.75:46205 failed Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: <core> [core/tcp_main.c:2997]: tcpconn_1st_send(): connect 10.45.6.75:37861 failed (RST) Connection refused Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: <core> [core/tcp_main.c:3005]: tcpconn_1st_send(): 10.45.6.75:37861: connect & send for 0x7fea3e2a3660 failed: Connection refused (111) Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: tm [../../core/forward.h:261]: msg_send_buffer(): tcp_send failed Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: tm [uac.c:644]: send_prepared_request_impl(): Attempt to send to precreated request failed Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=846 a=25 n=sht_iterator_next Apr 6 18:03:29 tel-VirtualBox kamailio[9730]: ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=856 a=25 n=sht_iterator_end Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=861 a=25 n=xlog Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: <script>: request sent to sip:bob@10.45.6.75:60060;rinstance=a59e664cda799366;transport=tcp completed with code: 408, Type 2 Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=888 a=16 n=if Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=868 a=16 n=if Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=865 a=63 n=assign Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=886 a=16 n=if Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=861 a=25 n=xlog Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: <script>: request sent to sip:bob@10.45.6.75:46205;transport=TCP;rinstance=f2285c01dd4acb41 completed with code: 408, Type 2 Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=888 a=16 n=if Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=868 a=16 n=if Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=865 a=63 n=assign Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=886 a=16 n=if Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=861 a=25 n=xlog Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: <script>: request sent to sip:bob@10.45.6.75:37861;transport=TCP;rinstance=e965f9424c56508e completed with code: 408, Type 2 Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=888 a=16 n=if Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=868 a=16 n=if Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=865 a=63 n=assign Apr 6 18:03:32 tel-VirtualBox kamailio[9719]: ERROR: *** cfgtrace:request_route=[uac:reply] c=[/etc/kamailio/pcscf/kamailio.cfg] l=886 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=444 a=25 n=xlog Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: <script>: REGISTER ( sip:alice@sip.example.com;transport=UDP (10.45.6.174:43244) to sip:alice@sip.example.com;transport=UDP, xJ8PR1SPmFBplNjHGYC1LA..) Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=455 a=5 n=route Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=668 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=663 a=25 n=is_method Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=668 a=63 n=assign Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=669 a=26 n=xlog Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=709 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=714 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=709 a=25 n=mf_process_maxfwd_header Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=720 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=714 a=26 n=sanity_check Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=725 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=720 a=24 n=has_totag Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=730 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=725 a=25 n=is_method Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=734 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=730 a=24 n=t_lookup_request Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=737 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=734 a=25 n=is_method Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=735 a=26 n=send_reply Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=466 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=458 a=25 n=is_method Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=472 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=466 a=25 n=is_method Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=472 a=5 n=route Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/pcscf/kamailio.cfg] l=788 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/pcscf/kamailio.cfg] l=748 a=24 n=has_totag Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=481 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=477 a=24 n=t_precheck_trans Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=481 a=24 n=t_check_trans Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=484 a=24 n=t_check_trans Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=490 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=486 a=25 n=is_method Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=495 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=490 a=25 n=is_method Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=495 a=24 n=loose_route Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=640 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header present. Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=585 a=55 n=force_rport Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=589 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=586 a=25 n=is_method Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=587 a=24 n=add_contact_alias Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=589 a=39 n=setflag Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=597 a=16 n=if Apr 6 18:04:19 tel-VirtualBox kamailio[9606]: ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=591 a=25 n=is_method Kindly do the needful Thanks.

1 0

Problem Storing USRLOC Attributes, Part 2
by Sergiu Pojoga 06 Apr '20

06 Apr '20

Hi there, I'm observing a similar behavior as reported a while ago in this case: https://github.com/kamailio/kamailio/issues/243 It appears that only the *last *attribute before save() is stored in DB and later restored on lookup() kamcmd version kamailio 5.2.5 (x86_64/linux) e10d4f Script: $xavp(ulattrs=>PubNetIP) = $si; $xavp(ulattrs=>PubNetPort) = $sp; $xavp(ulattrs=>Transport) = $proto; pv_xavp_print(); save("location", ""); Mysql log: insert into `location_attrs` (`username`,`ruid`,`last_modified`,`aname`,`atype`,`avalue`,`domain` ) values ('firstuser','uloc-5e855601-2529-1','2020-04-01 23:03:48','Transport',0,'udp','sip.mydomain.net') MariaDB [(none)]> select * from kamailio.location_attrs; *************************** 1. row *************************** id: 26 ruid: uloc-5e855601-2529-1 username: firstuser domain: sip.mydomain.net aname: Transport atype: 0 avalue: udp last_modified: 2020-04-01 23:12:48 Kamailio xavp pring in logs: +++++ start XAVP list: 0x7f28ff1e9590 ((nil)) (level=0) *** (l:0 - 0x7f28ff1e9590) XAVP name: ulattrs XAVP id: 2085639187 XAVP value type: 6 XAVP value: <xavp:0x7f28ff1e94e0> +++++ start XAVP list: 0x7f28ff1e94e0 (0x7f28ff1e95b0) (level=1) *** (l:1 - 0x7f28ff1e94e0) XAVP name: Transport XAVP id: 3694982711 XAVP value type: 2 XAVP value (str): udp *** (l:1 - 0x7f28ff1e94e0) end ----- end XAVP list: 0x7f28ff1e94e0 (level=1) *** (l:0 - 0x7f28ff1e9590) end *** (l:0 - 0x7f28ff1e9438) XAVP name: ulattrs XAVP id: 2085639187 XAVP value type: 6 XAVP value: <xavp:0x7f28ff1e9388> +++++ start XAVP list: 0x7f28ff1e9388 (0x7f28ff1e9458) (level=1) *** (l:1 - 0x7f28ff1e9388) XAVP name: PubNetPort XAVP id: 3296056264 XAVP value type: 1 XAVP value (int): 58535 *** (l:1 - 0x7f28ff1e9388) end ----- end XAVP list: 0x7f28ff1e9388 (level=1) *** (l:0 - 0x7f28ff1e9438) end *** (l:0 - 0x7f28ff1e92e0) XAVP name: ulattrs XAVP id: 2085639187 XAVP value type: 6 XAVP value: <xavp:0x7f28ff1e9228> +++++ start XAVP list: 0x7f28ff1e9228 (0x7f28ff1e9300) (level=1) *** (l:1 - 0x7f28ff1e9228) XAVP name: PubNetIP XAVP id: 3296017806 XAVP value type: 2 XAVP value (str): 173.246.28.83 *** (l:1 - 0x7f28ff1e9228) end ----- end XAVP list: 0x7f28ff1e9228 (level=1) *** (l:0 - 0x7f28ff1e92e0) end ----- end XAVP list: 0x7f28ff1e9590 (level=0)

2 4

Substitution of Contact Header
by Mack Hendricks 06 Apr '20

06 Apr '20

Hey All, I always have issues with substituting out header values. I’ve tried subst and subst_hf. Here is the value that I want to change out: Contact: <sip:52.41.52.34:5060;line=sr-h06Cy7RYRFkYacM0a7epa7sVa7sVRcNCW7IbaGZ9RGlC> And this is what I want it to be Contact: <sip:1313717555@jump.dsiprouter.net <sip:1313717555@jump.dsiprouter.net>;line=sr-h06Cy7RYRFkYacM0a7epa7sVa7sVRcNCW7IbaGZ9RGlC> I’m using this subst('/^Contact: <sip:(.*);(.*)>$/Contact: <sip:1313717555@jump.dsiprouter.net;\2>\r/ig’); And it doesn’t work, but putting into a online regular expression tester it looks good as shown below. Any ideas?

1 0

Confused with forwarding REGISTER
by fuxfwgc4a2i1gr 06 Apr '20

06 Apr '20

Hi. Well issue already described there : https://github.com/kamailio/kamailio/issues/2271 I will be really happy if somebody point me what exactly i have to do to route 404 answer from Registar to UA. If i understand right, in moment when registrar generate 404 i have to see in log message which generated inside onreply_route[REPLYROUTE] but in current moment i not see it and 404 not transmitted to UA. And it just silently ignored by Kamailio. In case if i am right and this is a bug in Kamailio please point me what words i have to use to prove that this is really bug. In current moment routing config looks like : request_route { # per request initial checks route(REQINIT); route(CATCH_CANCEL); route(RR_PARSE); route(REGISTRATION); xlog("L_INFO", "Initial relay packet\n"); $du = "sip:10.34.64.2:5060"; route(FORWARD); # update $du to set the destination address for proxying exit; } route[CATCH_CANCEL] { if (is_method("CANCEL") == true) { if (t_check_trans() == true) { route(FORWARD); } sl_reply("500", "Internal Server Error"); drop; } } route[RR_PARSE] { if(loose_route() == true ) { xlog("L_INFO", "RR-enforced\n"); append_hf("P-hint: rr-enforced\r\n"); route(FORWARD); } else { xlog("L_INFO", "RR simple\n"); record_route(); } } route[REGISTRATION] { if(is_method("REGISTER") == true) { add_path_received(); append_hf("Supported: path\r\n"); xlog("L_INFO", "GOT REGISTER\n"); } } # Per SIP request initial checks route[REQINIT] { if($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; } } route[FORWARD] { t_on_failure("FAILROUTE"); t_on_reply("REPLYROUTE"); if(t_relay() <= 0) { sl_reply_error(); } drop; } onreply_route[REPLYROUTE] { xlog("Reply SIP message from $si:$sp\n"); if (t_check_trans() == false) { drop; } } failure_route[FAILROUTE] { xlog("Reply Fail SIP message from $si:$sp\n"); }

4 6

Problem with max tcp connections
by Jose Fco. Irles Durá 06 Apr '20

06 Apr '20

Hi I have a kamailio 5.1.2 as load balancer and registration offloading, but I have a problem with the max tcp connections that it can handle. I suspect that is a linux limit, but I don't find the reason or config. When that limit arrives, I can't connect to kamailio and I receive "Connection reset by peer", but I can't view any error message in the logs. If I check the connections in kamailio, I view that it have "free" connections: # kamctl kamcmd core.tcp_info { readers: 8 max_connections: 4096 max_tls_connections: 2048 opened_connections: 2655 opened_tls_connections: 0 write_queued_bytes: 0 } I have this configs in kamailio.conf (related to tcp) disable_tcp=no tcp_connection_lifetime=3610 tcp_connect_timeout=5 tcp_crlf_ping=yes tcp_accept_aliases=no tcp_keepalive=yes tcp_keepidle=5 tcp_rd_buf_size=65536 tcp_conn_wq_max=131072 mlock_pages=yes shm_force_alloc=yes tcp_max_connections=4096 The shm memory to 256 and the pkg memory to 32. And, following this doc: https://github.com/kamailio/kamailio/blob/master/doc/tutorials/tcp_tunning.… I have setted this values: net.ipv4.ip_local_port_range = 1024 65535 net.core.somaxconn = 65535 net.core.netdev_max_backlog = 182757 Also, I had checked the limits for the main process pid: Limit Soft Limit Hard Limit Units Max cpu time unlimited unlimited seconds Max file size unlimited unlimited bytes Max data size unlimited unlimited bytes Max stack size 8388608 unlimited bytes Max core file size unlimited unlimited bytes Max resident set unlimited unlimited bytes Max processes unlimited unlimited processes Max open files 1048576 1048576 files Max locked memory 16777216 16777216 bytes Max address space unlimited unlimited bytes Max file locks unlimited unlimited locks Max pending signals 386297 386297 signals Max msgqueue size 819200 819200 bytes Max nice priority 0 0 Max realtime priority 0 0 Max realtime timeout unlimited unlimited us The service is running inside a lxc container, without any resource limit, connected to the outside word throught macvlan interface. Where can I find problem source? Best regards

4 13

need help with RTPProxy in bridged mode
by Grant Bagdasarian 04 Apr '20

04 Apr '20

Hello, I'm using RTPproxy for the first time in bridged mode and I can't get kamailio/rtpproxy to rewrite the c parameter to the correct public ip address of kamailio. The setup is as following: Carrier ------[fiber]------ Kamailio ---------[lan]--------- Freeswitch Kamailio is listening on two interfaces: 1) Private: 172.0.0.1 2) Public: 192.168.0.1 (since we have a dedicated fiber with our carrier, this is its public address) Freeswitch is listening on: 1) 172.0.0.2 Carrier is on: 1) 10.0.0.1 I've started an rtpproxy instance on the Kamailio box using: rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.1 172.0.0.1 I've played around with rtpproxy_manage() and the various flags (ie, ei), but I can't get kamailio to set the correct public IP when the 200 OK has to be sent back to the carrier. It always sets it to its private address, instead of its public address. I'm using Kamailio 4.2 with sippy/rtpproxy 2.0. Could someone please point me into the right direction? Thanks! Grant

4 8

need help with RTPProxy in bridged mode
by Artur Hovsepyan 04 Apr '20

04 Apr '20

hi, i'm also having the same issue, same setup, but instead of private IP i have real IP on my WAN interface. and regardless that i've used rtpproxy_manage("coeir")and rtpproxy_manage("coier")kamailio is still replying with my internal (sip+rtp) ip in the outgoing to WAN SDP. Can you please help to understand how do i need to manager rtp proxy to send WAN IP in SDP? when my WAN IP is private IP and my carrier is also having private IP from same network, the issue is not seen. Thanks in advance,

1 1

rtpproxy bridge external(real ip)/internal
by Artur Hovsepyan 03 Apr '20

03 Apr '20

Hey, This is my first post here, so I wanted a bit of help. Appreciating that in advance. i have following setup: carrier(asterisk with sip ip=rtp ip=real IP) <==> kamailio+rtpproxy(ext real ip, internal ip) <==> internal asterisk(sip ip != rtp ip, both internal ips). rtprpoxy setup/usr/local/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy -F -s udp:127.0.0.1 7722 -l kam_int_ip kam_ext_ip -m 10000 -M 20000 -L 30000 -d DBUG LOG kamailio listens on both external and internal ipsdispatcher list: 0 towards carrier SOCKET: udp:external_ip2 towards internal asterisk SOCKET: udp:internal_ip calls being routed with if(src_ip==carrier's real ip) sent to route "from_carrier" and if(src_ip==int_ast_sip_ip) sent to route "to_carrier" in the routes calls are dispatched with if(!ds_select_dst("0", "4")) and if(!ds_select_dst("2", "4")) rtpproxy_manage is done in "NATMANAGE" route: #checking if call is from carrier or to carrier:if(src_ip!=carrier_ext_ip) { if(nat_uac_test("8")) { rtpproxy_manage("coier"); } else { rtpproxy_manage("cor"); }}else { rtpproxy_manage("coeir");} so, this setup works, when i setup private IP addresses(from same network) on both carrier side(10.0.1.1) and external interface of kamailio(10.0.1.2) (accordingly changing the dispatcher file, rtpproxy and the kamailio.cfg). as per below: carrier(asterisk with sip ip=rtp ip=10.0.1.1) <==> kamailio+rtpproxy(ext ip=10.0.1.2, internal ip=10.0.2.1) <==> internal asterisk(sip ip != rtp ip, both internal ips). rtprpoxy setup/usr/local/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy -F -s udp:127.0.0.1 7722 -l 10.0.2.1 10.0.1.2 -m 10000 -M 20000 -L 30000 -d DBUG LOG kamailio listens on both external and internal ipsdispatcher list: 0 towards carrier SOCKET: udp:10.0.1.22 towards internal asterisk SOCKET: udp:10.0.2.1 calls being routed with if(src_ip==10.0.1.1) sent to route "from_carrier" and if(src_ip==172.16.21.21) sent to route "to_carrier" in the routes calls are dispatched with if(!ds_select_dst("0", "4")) and if(!ds_select_dst("2", "4")) rtpproxy_manage is done in "NATMANAGE" route: #checking if call is from carrier or to carrier:if(src_ip!=10.0.1.1) { if(nat_uac_test("8")) { rtpproxy_manage("coier"); } else { rtpproxy_manage("cor"); }}else { rtpproxy_manage("coeir");} once i replace the IP addresses of carrier and kamailio with real IPs, kamailio is starting to forward internal asterisk's sip and rtp IP addresses to carrier and carrier's sip/rtp IPs in the SDP and breaking the SIP flow. that is is not the case while private IPs are setup for both external and internal interfaces. in the later case it normally does the bridging and sends correct info in SDP messages. can someone help me please to understand why/how it is working correctly for private IPs and "misbehaving" in case of real IPs. thanks,Art

1 1

Query on force_send_socket
by Amarnath K 03 Apr '20

03 Apr '20

Hi All, I'm using Kamailio version 5.1.9. In cfg file have routing to send out SIP messages from Kamailio stack. In this routing logic added force_send_socket("tcp:192.168.40.10:5060"); forward() I'm expecting the send port should be from 5060, but it always pick random port to send out message. my requirement is to send SIP messages from 5060 port instead of picking random port. Regards, Amarnath

5 6

Kamailio release
by Virgílio Cunha 03 Apr '20

03 Apr '20

Hi, When is the scheduled release date of kamailio-5.3.4? Best Regards, Virgílio Cunha

1 0

[ot] lightweight sip app for auto-answering with play file and echo mode
by Daniel-Constantin Mierla 01 Apr '20

01 Apr '20

Hello, wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing. Of course I know that Asterisk and FreeSwitch (or even SEMS) can do that, but they have many dependencies, requiring quite some resources to run them, so I thought maybe someone here figured out different solutions, eventually cli based apps like pjsua or baresip. GUI apps for Linux are also fine if they can be configured for such behaviour. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

7 18

Invite transaction lifetime
by Duarte Rocha 01 Apr '20

01 Apr '20

Greetings, I have a strange call with a client in which he sends me an INVITE and i do the usual "100", "183" and "200 - OK" but the client never gets my provisional or final responses and, as such keeps retransmitting the first INVITE. For the first retransmissions Kamailio doesn't relay the INVITE which i think is the correct behaviour. However, after around 30s, a retransmission is processed and is relayed to a different destination, using "ds_select_dst()" from the DISPATCH module. For this to happen it had to fail the "t_check_trans()" function from TM module which tells me that the transaction from the first INVITE was gone. I went to search timers and lifetime definitions on TM module and found one "max_inv_lifetime" which tells me that "An INVITE transaction will be kept in memory for maximum: max_inv_lifetime+fr_timer(from the ACK to the final reply wait)+wt_timer". In this case : max_inv_lifetime is 180 seconds, fr_timer is 30 seconds and wt_timer is 5 seconds. It's sum is 215seconds and from what i can see, my transaction only had a lifetime of 30s. fr_timer is configured with 30s but it's definition is "Timer which hits if no final reply for a request or ACK for a negative INVITE reply arrives (in milliseconds)." and it mentions ACK, but in my case it's an ACK for a final reply that is missing and not an ACK for a negative reply. What other reasons might have caused this transaction to be terminated ? Best Regards

1 0

Re: [SR-Users] Use topos module to hide contact in OPTIONS request
by Enzo 01 Apr '20

01 Apr '20

2 1

Dialog: dlg_var and multi-branch call
by SamyGo 31 Mar '20

31 Mar '20

Hi All, Just curious to ask as how to save $dlg_var for a multi-leg/branched call. When I call a subscriber with multiple Contacts, and one branch answer while rest are CANCEL'd from that point onward I've seen that the dialog and its variables disappear from Kamailio memory and DB.Is this how its intended to be ? Here are the relevant debug line: DEBUG: dialog [dlg_hash.c:1266]: next_state_dlg(): dialog 0x7f7c1461e2e8 changed from state 2 to state 5, due event 4 (ref 2) DEBUG: dialog [dlg_handlers.c:574]: dlg_onreply(): dialog 0x7f7c1461e2e8 failed (negative reply) DEBUG: dialog [dlg_cb.c:271]: run_dlg_callbacks(): dialog=0x7f7c1461e2e8, type=4 DEBUG: dialog [dlg_handlers.c:1050]: dlg_set_tm_waitack(): registering TMCB to wait for negative ACK Best Regard, Sammy

1 0

Can MSRP module relay to multiple destinations?
by Sefa Cebe 31 Mar '20

31 Mar '20

Hello, I'm using Kamailio MSRP module as a MSRP relay, It works successfully when I send one-to-one MSRP messages. I am having trouble while sending msrp messages to multiple receivers. My question is that can I use this module to forward msrp messages not just one receiver, but multiple receivers? Can this module be used like MSRP switch? Thanks in advance.

2 3

Kamailio breaks RFC's 'Contact Header'
by Edward Romanenco 31 Mar '20

31 Mar '20

Description I'm running Kamailio 5.2.0, whenever I relay an invite via Kamailio, my original contact header is changed from the original: sip:+XXXXXXXXX@YYY.YYY.YYY.YYYY:5060;transport=udp;gw=netvision To sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*. Even when I set it correctly on my route, using: remove_hf("Contact"); append_hf("Contact: sip:$tU@YYY.YYY.YYY.YYY:5060;transport=udp;gw=netvision\r\n", "Contact"); It still ends up being modified. What can I do to keep the contact header as it is? SIP Traffic U YYY.YYY.YYY.YYY:5060 -> 81.24.193.248:5060 INVITE sip:+442033202609@81.24.193.248:5060 SIP/2.0. Record-Route: <sip:YYY.YYY.YYY.YYY;r2=on;lr=on;ftag=mp0S9yH11vryH;vsf=AAAAAAAAAAAAAAAAAAAAAAADCAEACRgPGB8AAy4xNjM->. Record-Route: <sip:2YYY.YYY.YYY.YYY;line=sr-.n274V8TlidQMVyUlg87vV8nlfDXlVHB9VD0.VHB9VDoLGtNRbH7ltlTEkyQl3jXEky1LNMoRkt5ukt5ukt5ukt5ukt5ukt5ukt5ukt5W8M5WktckoLuWAsduktTMm64pCAD>. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK53e9.308596b23ed683368534c9d609dce0f6.0. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKsr-kA2uvfsUlerJWtIylgCXvV8n4edQlgIUlgKQ4Vk74gI1.oJVjb2njburlgCXvV8n4edQlgIUlgKQ4HWsRbWPLVDX.GrXLcAHlcC74nZXKbTVpcHh4b6mMGZvJgLsLcJTkgtGu2PGgy**. Max-Forwards: 67. From: +18702935016 <sip:+18702935016@YYY.YYY.YYY.YYY>;tag=mp0S9yH11vryH. To: <sip:+442033202609@81.24.193.248>. Call-ID: !!:jbkfMgyHjbsDjblfjqAQlVldvgyHKolDlcIHlcknKVIdjcIT. CSeq: 18182498 INVITE. Contact: <sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*>. User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20180706T160334Z~de3df8dc0e~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 224. Remote-Party-ID: "+18702935016" <sip:+18702935016@YYY.YYY.YYY.YYY>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1585439298 1585439299 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ. s=FreeSWITCH. c=IN IP4 ZZZ.ZZZ

2 3

Re: [SR-Users] How to correctly answer a PnP-SUBSCRIBE message
by Jan-Hendrik Dörner 31 Mar '20

31 Mar '20

Hello Daniel, thank you for your reply. I changed the line mentioned to t_uac_send("NOTIFY", "$(ct{s.substr,1,0}{s.striptail,1})", "$su", "192.168.1.156","Call-ID: $hdr(Call-ID)\r\nSubscription-State: terminated;reason=timeout\r\nEvent: $hdr(Event)\r\nContent-Type: application/url\r\nFrom: $fU\r\nTo: $hdr(From)", "http://server.conf/settings-{mac} <http://server.conf/settings-%7Bmac%7D>"); the result is: 3) Call-ID is fine now ; Problem solved :-) 1) The Via IP is now changed to the IP and port of the server, but not to the IP of the phone. 2) The sending interface / IP is not correct: It now sends _only_ from the wrong IP !! As a result, it repeats sending the NOTIFY-message over and over again since the message does not get to the phone ... Please see the starting of the SIP-traffic below U 192.168.1.152:57283 -> 224.0.1.75:5060 #53 SUBSCRIBE sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.152:57283;rport. From: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=176419003. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>. Call-ID: 277773303(a)192.168.1.152 <mailto:277773303@192.168.1.152>. CSeq: 1 SUBSCRIBE. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Expires: 0. Accept: application/url. Contact: <sip:192.168.1.152:57283 <sip:192.168.1.152:57283>>. User-Agent: snom760/8.9.3.80. Content-Length: 0. . # U 192.168.1.156:5060 -> 192.168.1.152:57283 #54 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.152:57283;rport=57283;received=192.168.1.152. From: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=176419003. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=fb721cde02dffb6bb099728401306798.1182. Call-ID: 277773303(a)192.168.1.152 <mailto:277773303@192.168.1.152>. CSeq: 1 SUBSCRIBE. Server: kamailio (5.2.5 (x86_64/linux)). Content-Length: 0. . # U 172.23.56.1:5060 -> 192.168.1.152:57283 #55 NOTIFY sip:192.168.1.152:57283 <sip:192.168.1.152:57283> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bKee8f.0ea476a1000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=176419003. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-9700. CSeq: 10 NOTIFY. Call-ID: 277773303(a)192.168.1.152 <mailto:277773303@192.168.1.152>. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} <http://server.conf/settings-%7Bmac%7D> # U 172.23.56.1:5060 -> 192.168.1.152:57283 #56 NOTIFY sip:192.168.1.152:57283 <sip:192.168.1.152:57283> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bKee8f.0ea476a1000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=176419003. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-9700. CSeq: 10 NOTIFY. Call-ID: 277773303(a)192.168.1.152 <mailto:277773303@192.168.1.152>. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} <http://server.conf/settings-%7Bmac%7D> # U 172.23.56.1:5060 -> 192.168.1.152:57283 #57 NOTIFY sip:192.168.1.152:57283 <sip:192.168.1.152:57283> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bKee8f.0ea476a1000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=176419003. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-9700. CSeq: 10 NOTIFY. Call-ID: 277773303(a)192.168.1.152 <mailto:277773303@192.168.1.152>. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} <http://server.conf/settings-%7Bmac%7D> # U 192.168.1.152:57283 -> 224.0.1.75:5060 #58 SUBSCRIBE sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.152:57283;rport. From: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=1472545741. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>. Call-ID: 1886698335(a)192.168.1.152 <mailto:1886698335@192.168.1.152>. CSeq: 1 SUBSCRIBE. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Expires: 0. Accept: application/url. Contact: <sip:192.168.1.152:57283 <sip:192.168.1.152:57283>>. User-Agent: snom760/8.9.3.80. Content-Length: 0. . # U 192.168.1.156:5060 -> 192.168.1.152:57283 #59 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.152:57283;rport=57283;received=192.168.1.152. From: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=1472545741. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=fb721cde02dffb6bb099728401306798.1182. Call-ID: 1886698335(a)192.168.1.152 <mailto:1886698335@192.168.1.152>. CSeq: 1 SUBSCRIBE. Server: kamailio (5.2.5 (x86_64/linux)). Content-Length: 0. . # U 172.23.56.1:5060 -> 192.168.1.152:57283 #60 NOTIFY sip:192.168.1.152:57283 <sip:192.168.1.152:57283> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bKa0df.967a0433000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=1472545741. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-d106. CSeq: 10 NOTIFY. Call-ID: 1886698335(a)192.168.1.152 <mailto:1886698335@192.168.1.152>. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} <http://server.conf/settings-%7Bmac%7D> # U 172.23.56.1:5060 -> 192.168.1.152:57283 #61 NOTIFY sip:192.168.1.152:57283 <sip:192.168.1.152:57283> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bKa0df.967a0433000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=1472545741. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-d106. CSeq: 10 NOTIFY. Call-ID: 1886698335(a)192.168.1.152 <mailto:1886698335@192.168.1.152>. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} <http://server.conf/settings-%7Bmac%7D> # (If I change the IP-parameter to "172.23.56.1" the resulting sending IP is unchanged!) > Am 30.03.2020 um 16:17 schrieb Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>>: > > Hello, > > On 30.03.20 12:07, Jan-Hendrik Dörner wrote: >> Hallo, >> >> several telephones send a SIP-multicast SUBSCRIBE message to address 224.0.1.75 if they have no settings-URL configured and receive none over DHCP. >> >> So I configure my kamailio server, to respond to that multicast-request. >> But: Although mine is working, I do feel it could be improved, since the NOTIFY packet send out is not quite as it should be … >> >> Here is what I did (the 172.23.56.1 is my voip-vlan-address) >> >> listen=udp:172.23.56.1:5060 advertise 192.168.1.156:5060 >> listen=udp:192.168.1.156:5060 advertise 192.168.1.156:5060 >> listen=tcp:127.0.0.1:5060 >> >> mcast="eth0" >> listen=udp:224.0.1.75:5060 >> >> …. >> >> if ($Ri == "224.0.1.75"){ >> if(is_method("SUBSCRIBE")){ >> xlog("L_INFO", "Received pnp-event from IP $si from User-Agent $hdr(User-Agent) und User $fU\r\n" ); >> sl_send_reply("200", "OK"); >> t_uac_send("NOTIFY", "$(ct{s.substr,1,0}{s.striptail,1})", "", "","Subscription-State: terminated;reason=timeout\r\nEvent: $hdr(Event)\r\nContent-Type: application/url\r\nFrom: $fU\r\nTo: $hdr(From)", "http://server.conf/settings-{mac} <http://server.conf/settings-%7Bmac%7D>"); >> }else { >> xlog("L_INFO", "Drop message $rm from pnp-Interface \r\n" ); >> } >> exit; >> } >> >> >> And here are the SIP-packets I see on my server (192.168.1.156) coming from my snom-phone (192.168.1.152). >> My problems with the NOTIFY -packets are: >> 1) How do I set the Via parameters to be the same as in the OK-reply? > > Try to set the next hop address to "$su" and combined with 2) maybe it > results in what you want to get. > > >> 2) How do I send out the NOTIFY only via the 192.168.1.156 network interface? > > Set the local socket parameter from where to send the NOTIFY. > > >> 3) May I send out the NOTIFY with the Call-ID from the SUBSCRIBE message? How? > > > Provide the Call-Id haeder as part of the headers parameter: > > - > https://www.kamailio.org/docs/modules/stable/modules/tm.html#tm.f.t_uac_send <https://www.kamailio.org/docs/modules/stable/modules/tm.html#tm.f.t_uac_send> > > Cheers, > Daniel > > >> >> Any help is highly appreciated >> Jan-Hendrik >> >> # >> U 192.168.1.152:58714 -> 224.0.1.75:5060 #258 >> SUBSCRIBE sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan> SIP/2.0. >> Via: SIP/2.0/UDP 192.168.1.152:58714;rport. >> From: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=1750588964. >> To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>. >> Call-ID: 1395386659(a)192.168.1.152 <mailto:1395386659@192.168.1.152>. >> CSeq: 1 SUBSCRIBE. >> Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". >> Expires: 0. >> Accept: application/url. >> Contact: <sip:192.168.1.152:58714 <sip:192.168.1.152:58714>>. >> User-Agent: snom760/8.9.3.80. >> Content-Length: 0. >> . >> >> # >> U 192.168.1.156:5060 -> 192.168.1.152:58714 #259 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 192.168.1.152:58714;rport=58714;received=192.168.1.152. >> From: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=1750588964. >> To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=fb721cde02dffb6bb099728401306798.2f00. >> Call-ID: 1395386659(a)192.168.1.152 <mailto:1395386659@192.168.1.152>. >> CSeq: 1 SUBSCRIBE. >> Server: kamailio (5.2.5 (x86_64/linux)). >> Content-Length: 0. >> . >> >> # >> U 192.168.1.156:5060 -> 192.168.1.152:58714 #260 >> NOTIFY sip:192.168.1.152:58714 <sip:192.168.1.152:58714> SIP/2.0. >> Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0. >> To: <sip:MAC%3a0004137120D8@lan <sip:MAC%3a0004137120D8@lan>>;tag=1750588964. >> From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. >> CSeq: 10 NOTIFY. >> Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1 <mailto:24eaa8de01be1768-70960@172.23.56.1>. >> Max-Forwards: 70. >> Content-Length: 33. >> User-Agent: kamailio (5.2.5 (x86_64/linux)). >> Subscription-State: terminated;reason=timeout. >> Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". >> Content-Type: application/url. >> . >> http://server.conf/settings-{mac} <http://server.conf/settings-%7Bmac%7D> >> # >> U 172.23.56.1:5060 -> 192.168.1.152:58714 #261 >> NOTIFY sip:192.168.1.152:58714 SIP/2.0. >> Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0. >> To: <sip:MAC%3a0004137120D8@lan>;tag=1750588964. >> From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. >> CSeq: 10 NOTIFY. >> Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. >> Max-Forwards: 70. >> Content-Length: 33. >> User-Agent: kamailio (5.2.5 (x86_64/linux)). >> Subscription-State: terminated;reason=timeout. >> Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". >> Content-Type: application/url. >> . >> http://server.conf/settings-{mac} >> # >> U 172.23.56.1:5060 -> 192.168.1.152:58714 #262 >> NOTIFY sip:192.168.1.152:58714 SIP/2.0. >> Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0. >> To: <sip:MAC%3a0004137120D8@lan>;tag=1750588964. >> From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. >> CSeq: 10 NOTIFY. >> Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. >> Max-Forwards: 70. >> Content-Length: 33. >> User-Agent: kamailio (5.2.5 (x86_64/linux)). >> Subscription-State: terminated;reason=timeout. >> Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". >> Content-Type: application/url. >> . >> http://server.conf/settings-{mac} >> # >> U 192.168.1.152:58714 -> 224.0.1.75:5060 #263 >> SUBSCRIBE sip:MAC%3a0004137120D8@lan SIP/2.0. >> Via: SIP/2.0/UDP 192.168.1.152:58714;rport. >> From: <sip:MAC%3a0004137120D8@lan>;tag=1971605806. >> To: <sip:MAC%3a0004137120D8@lan>. >> Call-ID: 1846532696(a)192.168.1.152. >> CSeq: 1 SUBSCRIBE. >> Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". >> Expires: 0. >> Accept: application/url. >> Contact: <sip:192.168.1.152:58714>. >> User-Agent: snom760/8.9.3.80. >> Content-Length: 0. >> . >> >> # >> U 192.168.1.156:5060 -> 192.168.1.152:58714 #264 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 192.168.1.152:58714;rport=58714;received=192.168.1.152. >> From: <sip:MAC%3a0004137120D8@lan>;tag=1971605806. >> To: <sip:MAC%3a0004137120D8@lan>;tag=fb721cde02dffb6bb099728401306798.2f00. >> Call-ID: 1846532696(a)192.168.1.152. >> CSeq: 1 SUBSCRIBE. >> Server: kamailio (5.2.5 (x86_64/linux)). >> Content-Length: 0. >> . >> >> # >> U 192.168.1.156:5060 -> 192.168.1.152:58714 #265 >> NOTIFY sip:192.168.1.152:58714 SIP/2.0. >> Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bK201a.063839a4000000000000000000000000.0. >> To: <sip:MAC%3a0004137120D8@lan>;tag=1971605806. >> From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-f31b. >> CSeq: 10 NOTIFY. >> Call-ID: 24eaa8de01be1768-70963(a)172.23.56.1. >> Max-Forwards: 70. >> Content-Length: 33. >> User-Agent: kamailio (5.2.5 (x86_64/linux)). >> Subscription-State: terminated;reason=timeout. >> Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". >> Content-Type: application/url. >> . >> http://server.conf/settings-{mac} >> # >> U 192.168.1.152:58714 -> 192.168.1.156:5060 #266 >> SIP/2.0 200 Ok. >> Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0;received=192.168.1.156. >> From: <MAC:0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. >> To: <sip:MAC%3A0004137120D8@lan>;tag=1750588964. >> Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. >> CSeq: 10 NOTIFY. >> User-Agent: snom760/8.9.3.80. >> Content-Length: 0. >> . >> >> # >> U 192.168.1.152:58714 -> 192.168.1.156:5060 #267 >> SIP/2.0 200 Ok. >> Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0;received=192.168.1.156. >> From: <MAC:0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. >> To: <sip:MAC%3A0004137120D8@lan>;tag=1750588964. >> Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. >> CSeq: 10 NOTIFY. >> User-Agent: snom760/8.9.3.80. >> Content-Length: 0. >> . >> >> # >> U 192.168.1.152:58714 -> 192.168.1.156:5060 #268 >> SIP/2.0 200 Ok. >> Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bK201a.063839a4000000000000000000000000.0;received=192.168.1.156. >> From: <MAC:0004137120D8>;tag=da567df9df3ac234e43969a051db4641-f31b. >> To: <sip:MAC%3A0004137120D8@lan>;tag=1971605806. >> Call-ID: 24eaa8de01be1768-70963(a)172.23.56.1. >> CSeq: 10 NOTIFY. >> User-Agent: snom760/8.9.3.80. >> Content-Length: 0. >> . >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> > > -- > Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com/> > www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>

2 4

P-CSCF Error - help required asap
by Pavithra N M 30 Mar '20

30 Mar '20

Hi, I am facing the below error in my P-CSCF module as i am running P-CSCF, I-CSCF, S-CSCF, HSS in separate VMs. 0(3512) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(3512) ERROR: <script>: event_route[htable:mod-init] { 1(3513) ERROR: rtpengine [rtpengine.c:2839]: send_rtpp_command(): can't send command "ping" to RTP proxy <udp:localhost:9910> 1(3513) ERROR: rtpengine [rtpengine.c:2740]: rtpp_test(): proxy did not respond to ping 1(3513) ERROR: rtpengine [rtpengine.c:2839]: send_rtpp_command(): can't send command "ping" to RTP proxy <udp:localhost:9911> 1(3513) ERROR: rtpengine [rtpengine.c:2740]: rtpp_test(): proxy did not respond to ping 68(3637) INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) 72(3644) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing 73(3645) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing 69(3641) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing 75(3653) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing 71(3643) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/3643 77(3655) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing 76(3654) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing 74(3652) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing 79(3657) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing root@tel-VirtualBox:/etc/kamailio/pcscf# 78(3656) WARNING: ims_usrloc_pcscf [usrloc_db.c:67]: connect_db(): DB connection already open... continuing 69(3641) ERROR: *** cfgtrace:request_route=[NATPING] c=[/etc/kamailio/pcscf/kamailio.cfg] l=836 a=5 n=route 69(3641) ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=895 a=16 n=if 69(3641) ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=895 a=63 n=assign 69(3641) ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=897 a=27 n=sql_query 69(3641) ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] l=898 a=25 n=xlog 69(3641) ERROR: <script>: Preloading NAT-PING. Rows: 0 69(3641) ERROR: *** cfgtrace:request_route=[preload_pcscf] c=[/etc/kamailio/pcscf/kamailio.cfg] kindly help me in this issue of how to configure the rtp proxy or even if it is not required for basic configuration,let me know the procedure to comment this part or the procedure to take it forward to run P-CSCF without any errors. Kindly help me in this regard. Thanks Best Regards Pavithra M Tata Elxsi [cid:7ca5c450-5bb2-4d74-85ab-1c3e16346e2e] ________________________________ Disclaimer: 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 are not the intended recipient of this message , or if this message has been addressed to you in error, please immediately alert the sender by reply email and then delete this message and any attachments. If you are not the intended recipient, you are hereby notified that any use, dissemination, copying, or storage of this message or its attachments is strictly prohibited. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors, omissions or contaminations in the contents of this message which might have occurred as a result of email transmission. If verification is required, please request for a hard-copy version. ________________________________

1 0

How to correctly answer a PnP-SUBSCRIBE message
by Jan-Hendrik Dörner 30 Mar '20

30 Mar '20

Hallo, several telephones send a SIP-multicast SUBSCRIBE message to address 224.0.1.75 if they have no settings-URL configured and receive none over DHCP. So I configure my kamailio server, to respond to that multicast-request. But: Although mine is working, I do feel it could be improved, since the NOTIFY packet send out is not quite as it should be … Here is what I did (the 172.23.56.1 is my voip-vlan-address) listen=udp:172.23.56.1:5060 advertise 192.168.1.156:5060 listen=udp:192.168.1.156:5060 advertise 192.168.1.156:5060 listen=tcp:127.0.0.1:5060 mcast="eth0" listen=udp:224.0.1.75:5060 …. if ($Ri == "224.0.1.75"){ if(is_method("SUBSCRIBE")){ xlog("L_INFO", "Received pnp-event from IP $si from User-Agent $hdr(User-Agent) und User $fU\r\n" ); sl_send_reply("200", "OK"); t_uac_send("NOTIFY", "$(ct{s.substr,1,0}{s.striptail,1})", "", "","Subscription-State: terminated;reason=timeout\r\nEvent: $hdr(Event)\r\nContent-Type: application/url\r\nFrom: $fU\r\nTo: $hdr(From)", "http://server.conf/settings-{mac}"); }else { xlog("L_INFO", "Drop message $rm from pnp-Interface \r\n" ); } exit; } And here are the SIP-packets I see on my server (192.168.1.156) coming from my snom-phone (192.168.1.152). My problems with the NOTIFY -packets are: 1) How do I set the Via parameters to be the same as in the OK-reply? 2) How do I send out the NOTIFY only via the 192.168.1.156 network interface? 3) May I send out the NOTIFY with the Call-ID from the SUBSCRIBE message? How? Any help is highly appreciated Jan-Hendrik # U 192.168.1.152:58714 -> 224.0.1.75:5060 #258 SUBSCRIBE sip:MAC%3a0004137120D8@lan SIP/2.0. Via: SIP/2.0/UDP 192.168.1.152:58714;rport. From: <sip:MAC%3a0004137120D8@lan>;tag=1750588964. To: <sip:MAC%3a0004137120D8@lan>. Call-ID: 1395386659(a)192.168.1.152. CSeq: 1 SUBSCRIBE. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Expires: 0. Accept: application/url. Contact: <sip:192.168.1.152:58714>. User-Agent: snom760/8.9.3.80. Content-Length: 0. . # U 192.168.1.156:5060 -> 192.168.1.152:58714 #259 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.152:58714;rport=58714;received=192.168.1.152. From: <sip:MAC%3a0004137120D8@lan>;tag=1750588964. To: <sip:MAC%3a0004137120D8@lan>;tag=fb721cde02dffb6bb099728401306798.2f00. Call-ID: 1395386659(a)192.168.1.152. CSeq: 1 SUBSCRIBE. Server: kamailio (5.2.5 (x86_64/linux)). Content-Length: 0. . # U 192.168.1.156:5060 -> 192.168.1.152:58714 #260 NOTIFY sip:192.168.1.152:58714 SIP/2.0. Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan>;tag=1750588964. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. CSeq: 10 NOTIFY. Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} # U 172.23.56.1:5060 -> 192.168.1.152:58714 #261 NOTIFY sip:192.168.1.152:58714 SIP/2.0. Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan>;tag=1750588964. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. CSeq: 10 NOTIFY. Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} # U 172.23.56.1:5060 -> 192.168.1.152:58714 #262 NOTIFY sip:192.168.1.152:58714 SIP/2.0. Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan>;tag=1750588964. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. CSeq: 10 NOTIFY. Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} # U 192.168.1.152:58714 -> 224.0.1.75:5060 #263 SUBSCRIBE sip:MAC%3a0004137120D8@lan SIP/2.0. Via: SIP/2.0/UDP 192.168.1.152:58714;rport. From: <sip:MAC%3a0004137120D8@lan>;tag=1971605806. To: <sip:MAC%3a0004137120D8@lan>. Call-ID: 1846532696(a)192.168.1.152. CSeq: 1 SUBSCRIBE. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Expires: 0. Accept: application/url. Contact: <sip:192.168.1.152:58714>. User-Agent: snom760/8.9.3.80. Content-Length: 0. . # U 192.168.1.156:5060 -> 192.168.1.152:58714 #264 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.152:58714;rport=58714;received=192.168.1.152. From: <sip:MAC%3a0004137120D8@lan>;tag=1971605806. To: <sip:MAC%3a0004137120D8@lan>;tag=fb721cde02dffb6bb099728401306798.2f00. Call-ID: 1846532696(a)192.168.1.152. CSeq: 1 SUBSCRIBE. Server: kamailio (5.2.5 (x86_64/linux)). Content-Length: 0. . # U 192.168.1.156:5060 -> 192.168.1.152:58714 #265 NOTIFY sip:192.168.1.152:58714 SIP/2.0. Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bK201a.063839a4000000000000000000000000.0. To: <sip:MAC%3a0004137120D8@lan>;tag=1971605806. From: <MAC%3a0004137120D8>;tag=da567df9df3ac234e43969a051db4641-f31b. CSeq: 10 NOTIFY. Call-ID: 24eaa8de01be1768-70963(a)172.23.56.1. Max-Forwards: 70. Content-Length: 33. User-Agent: kamailio (5.2.5 (x86_64/linux)). Subscription-State: terminated;reason=timeout. Event: ua-profile;profile-type="device";vendor="snom";model="snom760";version="8.9.3.80". Content-Type: application/url. . http://server.conf/settings-{mac} # U 192.168.1.152:58714 -> 192.168.1.156:5060 #266 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0;received=192.168.1.156. From: <MAC:0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. To: <sip:MAC%3A0004137120D8@lan>;tag=1750588964. Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. CSeq: 10 NOTIFY. User-Agent: snom760/8.9.3.80. Content-Length: 0. . # U 192.168.1.152:58714 -> 192.168.1.156:5060 #267 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bKf21c.917eaf36000000000000000000000000.0;received=192.168.1.156. From: <MAC:0004137120D8>;tag=da567df9df3ac234e43969a051db4641-8cfb. To: <sip:MAC%3A0004137120D8@lan>;tag=1750588964. Call-ID: 24eaa8de01be1768-70960(a)172.23.56.1. CSeq: 10 NOTIFY. User-Agent: snom760/8.9.3.80. Content-Length: 0. . # U 192.168.1.152:58714 -> 192.168.1.156:5060 #268 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 224.0.1.75;branch=z9hG4bK201a.063839a4000000000000000000000000.0;received=192.168.1.156. From: <MAC:0004137120D8>;tag=da567df9df3ac234e43969a051db4641-f31b. To: <sip:MAC%3A0004137120D8@lan>;tag=1971605806. Call-ID: 24eaa8de01be1768-70963(a)172.23.56.1. CSeq: 10 NOTIFY. User-Agent: snom760/8.9.3.80. Content-Length: 0. .

2 1

setting $du in branch route has no effect in 5.3
by Juha Heinanen 30 Mar '20

30 Mar '20

I noticed that in 5.3 and master setting $du in branch route has no effect. It works fine in 5.2 using exactly the same config. -- Juha

2 12

Problem faced when Using Kamaiio as Session Refresher
by harneet singh 30 Mar '20

30 Mar '20

Hi, I am fairly new to Kamailio and had a question regarding how to use Kamailio to enable Session refresh functionality when Kamailio is acting as Sip Stateful Proxy. Kamailio Version used: *5.3.2* with *Call Load based routing* using the *dispatcher *module. * From what i understand from the documentation, Kamailio will probably not be acting as a session refresher, but Kamailio can tear down the call in case session refresh is negotiated, between the caller and the callee(via Kamailio Sip Proxy), and no message exchange happens in the duration set in Session-Expires header. *Is my understanding correct?* ** *I believe the above functionality is possible by using the *sst* and *dialog* module. I have set the same according to the documentation but I keep getting the following error: "dialog [dlg_handlers.c:681]: *get_dlg_timeout(): invalid AVP value, using default timeout*" Can someone share a working example? * When i tried hardcoding the timeout value by setting the timeout_avp to a specific value, Kamailio did sense a timeout and hence sent a BYE towards the caller and the Callee side both(which is what the requirement is), however, i do see the *dialog is still not cleared* in the "kamcmd dispatcher.list". Output excerpt below for reference: [root@CPaaSVM ~]# kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:172.27.44.121:5080 ;transport=tcp FLAGS: AP PRIORITY: 0 ATTRS: { BODY: duid=sample-cas;maxload=1000 DUID: sample-cas MAXLOAD: 1000 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 111.304000 STD: 1042.193000 EST: 2.385000 MAX: 9999 TIMEOUT: 1 } RUNTIME: { DLGLOAD: *1* } } } } } } It is noteworthy that in case the BYE is initiated by either the caller or the callee, the dialog is cleared properly and the DLGLOAD is set to 0 on call termination. Any pointers for the above questions would be highly appreciated. Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

3 17

Transaction Matching Failed - ACK BYE
by egemen ulus 29 Mar '20

29 Mar '20

Hi All, I'm trying to establish a topology with Kamailio which is located between Asterisk and Outside. My issue is "transaction matching failed" (t_check_trans failed). That means Kamailio does not recognise ACK & BYE message to forwards. I had a check on email list and google but found nothing. I'm using it with record_route. SIP signalling and part of kamailio.cfg is below. Is there any suggestion? route[WITHINDLG] { if (!has_totag()) return; 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") ) { xlog("ack is came: $si:$sp\n"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not here"); exit; } route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } route(RTPPROXY); if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; 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; } Asterisk > Kamailio INVITE sip:user@outside_ip SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK1b56cce0 Max-Forwards: 70 From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip> Contact: <sip:0553847<tel:+10553847>aaaa@asterisk_ip:5060> Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.32.3 Kamailio > Outside INVITE sip:user@outside_ip SIP/2.0 Record-Route: <sip:kamailio_ip;lr;nat=yes> Via: SIP/2.0/UDP kamailio_ip;branch=z9hG4bK7b3f.8e4716eeea4b0dc45cc51d21b5aded57.0 Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0 From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip> Contact: <sip:0553847<tel:+10553847>aaaa@asterisk_ip:5060;alias=asterisk_ip~5060~1> Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.32.3 Outside > Kamailio SIP/2.0 200 OK Via: SIP/2.0/UDP kamailio_ip;branch=z9hG4bK7b3f.8e4716eeea4b0dc45cc51d21b5aded57.0 Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0 From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip>;tag=kv8iml6ur9 Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 INVITE Record-Route: <sip:kamailio_ip;lr;nat=yes> Contact: <sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp> Kamailio > Asterisk SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0 From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip>;tag=kv8iml6ur9 Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 INVITE Record-Route: <sip:kamailio_ip;lr;nat=yes> Contact: <sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp> Asterisk > Kamailio THIS ACK IS NOT ROUTED TO OUTSIDE BY KAMAILIO!! ACK sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK73366924 Route: <sip:kamailio_ip;lr;nat=yes> From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip>;tag=kv8iml6ur9 Contact: <sip:0553847<tel:+10553847>aaaa@asterisk_ip:5060> Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.32.3 Get Outlook for Android<https://aka.ms/ghei36>

2 6

Kamailiio TLS and Let'sEncrypt certs
by Igor Olhovskiy 29 Mar '20

29 Mar '20

Hi! Actually I’m trying to get Kamailio to work as MS Teams SBC following by perfect article https://skalatan.de/en/blog/kamailio-sbc-teams <https://skalatan.de/en/blog/kamailio-sbc-teams> It works well, but one thing is bothering me. I’m using Let’sEncrypt certs (actually, works well), but with setting in tls.conf verify_certificate = yes require_certificate = yes It’s giving an errors like /usr/sbin/kamailio[4551]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:1416F086:SSL routines:tls_process_server_certificate:certificate verify failed /usr/sbin/kamailio[4551]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f03e6d23d88 r: 0x7f03e6d23e08 (-1) They are resolved with setting these settings (verify/require) to off (actually, as mentioned here - https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ <https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/>), but I’m really curious - why? As I got, it’s using openssl verify on a background, but this check locally passed with openssl verify -CAfile issuer.crt myserver.crt myserver.crt: OK So, is there any tricks to lets encrypt or just some misconfig in tls.cfg? Now it looks like one from article [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/tls/myserver.key certificate = /etc/kamailio/tls/myserver.crt ca_list = /etc/kamailio/tls/issuer.crt [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/tls/myserver.key certificate = /etc/kamailio/tls/myserver.crt ca_list = /etc/kamailio/tls/issuer.crt — Regards, Igor

3 3

Kamailio OPTIONS Round-Trip
by Soltanici Ilie 27 Mar '20

27 Mar '20

Hi, With Asterisk, we are able to get some peer round-trip connection statistic by setting qualify=yes for the specified peer. It sends periodic OPTIONS to the peer and calculates the time round trip time. It's something like - "Status: OK (30 ms)". Is there any way to achieve this in Kamailio by using nathelper module, or any other? Thank you.

9 13

Segmentation fault in DMQ module
by Virgílio Cunha 27 Mar '20

27 Mar '20

Hello, I have in production environment with 4 proxies using DMQ to sync terminal registers and the DMQ configuration parameters is following: loadmodule "dmq.so" #modparam("dmq", "server_address", "sip:MY_IP_ADDRESS:MY_PORT_ADDRESS") modparam("dmq", "notification_address", "DMQ_HOSTS") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) In the lab environment with the same configuration that in prod, I can simulate this problem with 12000 terminal REGISTERS and, when doing several restarts on a proxy, one of them is generated a coredump. Analyzing the core, I can see that the problem is concurrency in the dmq synchronization. Job_queue_size (...) is called with a valid *queue *and, but inside this function, the *queue *is null. (gdb) up #1 0x00007fefe58ee246 in job_queue_size (queue=0x0) at worker.c:254 254 return atomic_get(&queue->count); (gdb) print queue $1 = (job_queue_t *) 0x0 (gdb) up #2 0x00007fefe58ed920 in add_dmq_job (msg=0x7ff028b453f8, peer=0x7fefebcc4df8) at worker.c:184 184 if(job_queue_size(workers[i].queue) == 0) { (gdb) print i $2 = 1 (gdb) print workers[i].queue $3 = (job_queue_t *) 0x7fefebce49b0 If I configure DMQ with num_workers=1 I can't reproduce this problem. I'm using the kamailio 5.0.8 release. Is this problem known to you? What is the right way to solve it? Best Regards, Virgílio Cunha

3 7

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