- sr-users - kamailio.org

secfilter table error
by Social Boh 07 May '20

07 May '20

Hello, any help with this error? ERROR 1067 (42000) at line 1: Invalid default value for 'action' ERROR: Creating extra tables failed at secfilter! Kamilio version 5.3.3 compiled from source. Thank you -- --- I'm SoCIaL, MayBe

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Kamailio v5.3.4 Released
by Daniel-Constantin Mierla 06 May '20

06 May '20

Hello, Kamailio SIP Server v5.3.4 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.4. For more details about version 5.3.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/05/kamailio-v5-3-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Calls from Kamailio to Teams
by sip user 06 May '20

06 May '20

Hi, I have connected Kamailio like SBC with Teams, and Calls from Teams to Kamailio works.. Now, I have problems with calls from Kamailio to Teams, not works. How have I send the call? I set record_route_preset, but how send the call to Teams? Thanks so much

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Releasing Kamailio v5.3.4
by Daniel-Constantin Mierla 06 May '20

06 May '20

Hello, based on the discussions from yesterday's online devel meeting, I am planning to release next week (likely on Wednesday, May 6) the next minor version out of branch 5.3, respectively 5.3.4. Should anyone be aware of issues not reported to the bug tracker or commits for fixes in master not backported to 5.3 branch, let us know. For new issues, the best is to report them on the tracker. For commits, you can provide links/hashids or make pull requests. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Map Diversion to History-Info
by Duarte Rocha 05 May '20

05 May '20

Greetings, Does Kamailio have any method to transform an History-Info with redirecting info into a Diversion or the other way around? Best Regards

2 1

PUA and db_redis
by Carsten Bock 05 May '20

05 May '20

Hi, has anyone been using db_redis with the Presence/PUA modules? If yes, can you share your experience and the proper key definition? Thanks, Carsten -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

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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

2 2

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.

1 0

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!!!

1 0

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

1 0

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.

2 1

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

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