- sr-users - kamailio.org

Need help on how to implement presence_xml
by Michel Pelletier 25 Jun '24

25 Jun '24

Hello, I need to implement Event: presence, application/pidf+xml as described in RFC-3856. I understand I need to use the presence_xml module and generate PUBLISH requests to update the information and ensure that the NOTIFY get sent. I know nothing abour XCAP or wether I need it or not. Any information source out there that could help me? Regards, Michel Pelletier

1 0

TLS: client doesn't check certificate CN
by Marat Gareev 20 Jun '24

20 Jun '24

Hello, I noticed that Kamailio can route either to IP and FQDN. This means the server certificate CN is not checked by the client. How to enable something like the 'verify_peer' option? Thanks

1 0

Possibility to print values of Core Keywords in Kamailio configuration file
by sadik.oualla.mohamed＠gmail.com 20 Jun '24

20 Jun '24

Hello everyone, I have a question regarding the usage of core keywords, specifically the possibility of printing their values in log messages. While I understand that core keywords such as 'dst_port' and 'af' etc... can be accessed directly in if conditions, I am interested to know if there is a way to print the values of these core keywords for logging purposes. Thank you.

4 6

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Wed, 19 Jun 2024 11:54:05 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > What you should expect using this method is blocking while the exec runs > which could run you out of child processes while they complete. I'd also > expect to see an "unusual amount of CPU activity" which will be the result > of the forked shells being created and destroyed. > > All of that being said, yes it works. It's architecturally overly simple > and very inefficient. If your load is tiny, it might be ok. Anything > moderate like Henning and others have said, will use an inordinate amount > of cpu. I would not personally risk it. > > Thank you Brett, Henning and Alex Sadly the load won't be tiny and cannot risk it. For now I'll deploy a local nginx and fpm to run the scripts, which are in php, and will use the http_async method. Sadly we have to "http all the things" nowadays. Let's see if I'm able to write down all the logic tomorrow and do some testing. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

Adding custom headers to HTTP server response xhttp
by Sasivarunan 19 Jun '24

19 Jun '24

HI Team, I use tls and xhttp module to enable websocket interface for webrtc client to connect. I wanted to add "Strict-Transport-Security" header to xhttp response to add security. But currently I don't see a way to add it anywhere. I can see only server_header as an option, But this will get added to SIP messages as well. It 'd be great if anybody has a suggestion around this. Thanks Varunan

2 2

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Tue, 18 Jun 2024 13:54:41 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > Just want to add that exec is heavy and slow. I would not recommend it. > > Is there a reason you want to do this over http-ifying your script and > using async? > > Not really. The scripts are provided. Maybe I'm against "http all the things" which seems to be what everyone does nowadays. Why would http client be lighter than executing a script? That would be a design problem, right? -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

exec concurrency
by Jon Bonilla (Manwe) 18 Jun '24

18 Jun '24

Hi all I've been working with async http client and async db queries in the past but now I have to execute scripts and store the return values in avps. Since I have no experience with that, I wonder how to achieve concurrency with that scenario. Is there a way to exec async or suspend until the exec has finished and resume later? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 2

Psresence dialoginfo question
by Stefan Mititelu 17 Jun '24

17 Jun '24

Hello, I am doing BLF but I want different users to be able to setup different dialog info initial states. My question is if there's an already implemented way for doing that? I've looked over docs and C code so far and this is what I've found: 1. modparam("presence_dialoginfo", "force_dummy_dialog", 1) => will force sending initial NOTIFY with dialog state "terminated" 2. Function calls: dlginfo_agg_nbody()->dlginfo_agg_nbody_empty() which will build the pre-defined xml body with that "terminated" state. If not implemented so far, maybe this can be used to set a different initial state for a user (e.g. export a cfg function or take that info from DB?) Any opinions on this? Thank you, Stefan

2 2

db_mysql startup warnings
by Juha Heinanen 17 Jun '24

17 Jun '24

When I start Kamailio master version that has been build on Debian 12 with default-libmysqlclient-dev depending on libmariadb-dev-compat, I get several warnings like this: 2024-06-17T14:48:27.702827+03:00 siika /usr/bin/sip-proxy[297246]: WARNING: db_mysql [km_my_con.c:187]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring How can I prevent them? -- Juha

1 1

How to Check Source Port in Application Layer with Kamailio
by sadik.oualla.mohamed＠gmail.com 16 Jun '24

16 Jun '24

Hello kamailio lovers, I am currently working on a Kamailio configuration where I need to check the source port of incoming SIP messages. Initially, I used the $sp pseudo-variable to get the source port, which worked perfectly while testing with SIPp. Here is the snippets code that I use " if ($sp != 5080 && $sp != 8080 && $sp != 5060){ sl_send_reply(403, "forbidden"); xlog("this $rm received, is forbidden\n"); exit; } " However, I encountered an issue when I switched to using sipsak for testing. It seems that sipsak uses a randomly assigned port for the actual network connection, even though it sets the source port in the SIP headers (e.g., Via, To and Contact headers) to a specified port using the -l option. Here is the sipsak command that I use "sipsak -vv -s sip:jack@192.168.1.2:5060-l 5080" My goal is to have Kamailio check the source port as specified in the application layer, rather than the transport layer port (i.e., the ephemeral port assigned by the OS randomly like 54691, 33017...). Could anyone guide me on how to properly configure Kamailio to extract and check the port from the application layer or SIP headers, or am I doing this in the wrong way? Any advice or examples would be greatly appreciated. "I am not a native english speak, so I use chatGPT to make my questions clear" Thank you!

1 0

topos module issue on hangup
by Arun K R 13 Jun '24

13 Jun '24

Hello, I am using topos module to hide Record-Route and Route header as per the requirement from carrier. But the topos module hides, contact, via and routing headers also. Due to this reason when I am getting bye from the carrier I am not able to route the bye back to the agent. Bye from Agent to carrier is working fine. Bye from carrier to my internal network interface is not routing to client through public interface. I assume this may be because Via / Contact header has been hidden and changed by the topos module with the invite please help me for this Thanks Arun

3 5

Kamailio v5.8.2 Released
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, Kamailio SIP Server v5.8.2 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.2. For more details about version 5.8.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/06/kamailio-v5-8-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.8.2
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, I am considering to release Kamailio v5.8.2 (out of branch 5.8) on Wednesday, June 12, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 2

New features in v.5.8
by Leonid Fainshtein 10 Jun '24

10 Jun '24

Hi, In the "Kamailio v5.8.0 Release Notes" I see: "SIP overload control support for server (RFC7339)". Where can I find some more details about the feature? Also, the similar question about the new "influxdbc" module. What are the possible use cases for this module? Thank you, Leonid Fainshtein Xorcom Ltd

2 2

handling 300 redirect requests
by Rob Green 09 Jun '24

09 Jun '24

I am trying to proxy a request from server A to server C with me being server B. i send the invite from A to B, then i need to send to server R which returns back to me a 300 redirect, which i then need to take the answer from the Contsct and then proxy the original invite from A to C. server A can not handle 3xx responses and must be proxied. I havent been able to find any examples where the 300 is intercepted and a new invite to the target is sent. i have the following code but it results in no new INVITE and the original session is hung until it is canceled. Any idea what i may be doing wrong? In the example below i am just hardcoding the uri for the new invite instead of getting from the Contact as a debug exercise. I am using Kamailio 5.7.3 route[HANDLEREDIRECT] { xlog("L_INFO", "DBG: HANDLEREDIRECT"); msg_apply_changes(); $var(test) = $hdr(Contact); xlog("L_INFO","DBG: redirect headers [$var(test)]: from=[$fu] from_user=[$fU], to=[$tu] to_user=[$tU]"); # send to gateway $var(new_uri) = "sip:" + $tU + "@" + "gateway"; if ($var(new_uri) != "") { xlog("DBG: Redirecting to: $var(new_uri)\n"); $ru = $var(new_uri); $tu = $ru; record_route(); route(RELAY); exit; # i have also tried using #t_newtran(); # and #t_relay(); } else { xlog("DBG: No contact header found in 300 response\n"); } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("DBG: new branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "DBG: manage branch"); route(NATMANAGE); return; } onreply_route[MANAGE_REPLY] { xdbg("DBG: incoming reply\n"); xlog("L_INFO", "DBG: manage reply"); if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: got 3xx redirect"); route(HANDLEREDIRECT); drop; } return; } I also tried using uac_redirect module, but it stalls the incoming invite also, and i cant figure out how to modify the uri from the contact before sending (to add params) # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: following 3xx redirect"); get_redirects("3:1"); t_relay(); exit; } } Rob Green Chief Information Officer (CIO) Tel: 1-352-218-1490 Cell: 1-407-620-0932 Email: rob.green(a)nuwave.com Web: www.nuwave.com This communication is the Property of NUWAVE Communications and may contain confidential or privileged information. Unauthorized use of this communication is strictly prohibited and may be unlawful. If you have received this communication in Error, Please immediately notify the sender by reply e-mail and destroy all copies of the communication and any attachments.

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Setting Key/Value Pairs at Dialogue Level
by Pavan Kumar 08 Jun '24

08 Jun '24

Hi everyone, I am creating a dialogue when I receive INVITE. Is there a way to set key/value pair like attributes at dialog level, so that, when I receive BYE message I can retrieve that information and do some processing? Thank you, Pavan Kumar

2 2

Recommendation for a signaling B2BUA to work with Kamailio on Ubuntu
by Benoît Panizzon 07 Jun '24

07 Jun '24

Hi I wonder if anyone could maybe recommend what to use as signaling B2BUA to work with Kamailio to handle spiralling / looping calls and to reduce the number of Route, Via Header and message size, which cause issue with a large number of CPE. Requirements: * New CallID needs to be generated * Handling Route and Via * ~ 2000 concurrent calls. * Preserving custom SIP X- header * Routing NOTIFY for MWI etc. Handling RTP is not required! What I have been looking at more closely: = FreeSwitch = Got stuck trying to solve the dependencies to get it to run/build on ubuntu jammy. Has anyone experience in running it in a docker container? Will it perform? = Sippy = Very small, very simple, unfortunately removes customer SIP headers and does not forward NOTIFY. (It's python, so maybe this could be easily implemented/fixed). = SEMS = Does not seem to be actively developed in the last, couple of years. Looks quite bloated to me for what I need. = Asterisk = Each customer header has to be copied, wants to handle RTP and bridge by issuing Re-Invites. Not sure how to route stuff like NOTIFY. Also from experience I fear asterisk would translate sip reply codes (like 608 and 607) it does not know to some 4XX default. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 3

Filtering codes in db acc
by Jon Bonilla (Manwe) 06 Jun '24

06 Jun '24

Hi all I'm trying to make kamailio not to insert a row in missed_calls table when reply code is 407. But even if I set the "failed_filter" modparam the 407 row is inserted. The acc mmodparams are: modparam("acc", "db_table_acc", "kam_acc") modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACC_MISS) modparam("acc", "db_url",DBURL) modparam("acc", "db_insert_mode", 1) modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "failed_filter", "401,407") modparam("acc", "db_extra", "whatever" db is mysql. Kamailio version is 5.7.4 When I send a call that receives a 407 and after auth a 480 y get both rows in missed_calls table. Log: DEBUG: acc [acc_mod.c:404]: mod_init(): acc db table initialized to: kam_acc DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 401 DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 1 = 407 DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cec807480) event type 512, reply code 407 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 407 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cebf59bc0) event type 512, reply code 486 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 486 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off am I configuring this wrong? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 3

CentOS 7 and 8 deprecated
by Sergey Safarov 06 Jun '24

06 Jun '24

The CentOS project page shows that the 7 and 8 releases are deprecated. https://www.centos.org/ Does the Kamailio community use these releases for prod? If yes then I will keep the packaging for this release. If not, I will remove the packaging for 7 and 8 releases, and in the future Kamailio will be packaged for 9 and 10 releases (planned). https://www.phoronix.com/news/CentOS-Stream-10-Start https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre… https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre…

2 1

Assistance Needed: Converting 11-Digit US Numbers to E.164 in Kamailio
by Pavan Kumar 04 Jun '24

04 Jun '24

Hey everyone, I have the following setup: External User/Trunk <===> Kamailio <===> FreeSWITCH I have configured FreeSWITCH and Kamailio to work only with phone numbers in E.164 format. Recently, I needed to integrate a new SIP trunk that sends phone numbers in an 11-digit US format. My initial idea to solve this is to use Kamailio as a translator that converts 11-digit numbers to E.164 when sending to FreeSWITCH, and back to 11-digit when sending to the trunk. Before writing the code, I want to know if there is already a solution to this problem in one of the modules, so that I won't reinvent the wheel. Thanks, Pavan Kumar

7 11

DMQ_USRLOC and multiple domains
by Lợi Đặng 03 Jun '24

03 Jun '24

Hi, I'm setting up shared user location for 2 instances of kamailio via dmq_usrloc. Everything seems fine except my setup is multi-domains. The issue arises when the producer node produces the Contact of domain X, Contact of domain Y separately, but the consumer node stores both as the Contact of domain Y (seen in kamcmd ul.dump), in which Y is set in usrloc_domain. modparam("dmq_usrloc", "usrloc_domain", "{{domain_Y}}") modparam("usrloc", "use_domain", 1) ==== I had a quick ngrep and see the domain was put in the "aor" key of the body in KMQ message for usrloc {"action":1,"aor":"test_username_0dppxcl@$domain_X"} But I'm not sure how to make my consumer node to store the contact in appropriate domain. Any help will be appreciated. Thanks and regards, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

1 1

ACK and Bye not relaying when record_route is not used
by Arun K R 03 Jun '24

03 Jun '24

Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call) My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side. Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: <sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11> Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248 v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101 ------------------------------------------------------------------------------------------------------- The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0 Thanks Arun

3 3

StriShaken module compilation
by Sergio Charrua 01 Jun '24

01 Jun '24

Hello all! I am struggling on compiling StriShaken module on RHEL 9.2. So far, the module was compiled as well as libstirshaken. Below are the steps used to compile (as sudo) both library and module (I hope this may help someone) and, of course, correct me if I am wrong in any step or if there is a better way. Also, note that the RHEL 9,2 is in a VM environment without access to the WWW, instead it uses Red Hat Satellite to download packages (via Yum or DNF), if available. Lib LibKS download libks from https://github.com/signalwire/libks unzip move to dir created by Unzip run yum groupinstall "Development Tools" dnf install libuuid-devel libatomic openssl-devel cmake . make make install cp /usr/lib/pkgconfig/libks2.pc /usr/lib64/pkgconfig/. For module LibStirShaken: download https://github.com/signalwire/libstirshaken unzip move to dir if using OpenSSL3.0 or + edit file configure.ac and add after line 28: if test x$HAVE_OPENSSL = x1; then openssl_CFLAGS="$openssl_CFLAGS -DHAVE_OPENSSL"; /* PATCH FOR OPENSSL3 */ AC_MSG_CHECKING([for OpenSSL >= 3.0.0]) AC_COMPILE_IFELSE([AC_LANG_PROGRAM([[ #include <openssl/opensslv.h> #if OPENSSL_VERSION_PREREQ(3,0) #error "you_have_version_3" #endif ]], [[]])], [ AC_MSG_RESULT([no]) ], [ AC_MSG_RESULT([yes]); AC_DEFINE(OPENSSL_SUPPRESS_DEPRECATED, 1, [disable openssl deprecated-function warnings]) ]) /* END OF PATCH */ else AC_MSG_ERROR([OpenSSL >= 1.0.1e and associated developement headers required]) fi run configure make sudo make install next, move to dir /root/kamailio/kamailio-5.7.4/src/modules/stirshaken cd /root/kamailio/kamailio-5.7.4/src/modules/stirshaken ln -s /root/stirshakenlib/libks-master/src/include/libks libks make make install All seem to compile OK , *but* when starting Kamailio I get the following errors: May 29 17:42:02 kamailio1 kamailio[248991]: ERROR: <core> [core/sr_module.c:599]: ksr_load_module(): could not open module </usr/local/lib64/kamailio/modules/stirshaken.so>: libstirshaken.so.1: cannot open shared object file: No such file or directory May 29 17:42:02 kamailio1 kamailio[248991]: CRITICAL: <core> [core/cfg.y:3915]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_stir_shaken_mod.cfg, line 106, column 12-26: failed to load module I haven't found a solution for this. Can anyone help? Thanks in advance! *Sérgio Charrua*

2 1

Parsing Error with SIP 607 Response
by Ghulam Mustafa 31 May '24

31 May '24

Hello Kamailio Community, I'm encountering a parsing error in Kamailio when receiving a SIP 607 "Unwanted" response. The relevant log entries are: May 31 12:32:09 k-proxy1 /usr/sbin/kamailio[320898]: ERROR: <core> [core/receive.c:378]: receive_msg(): core parsing of SIP message failed (36.53.xx.xx:5060/1) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 11) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<SIP/2.0 607#015#012Via: SIP/2.0/UDP 134.195.195.40:5060;branch=z9hG4bKc87a.ed6403ae1529f36b65171bf964604be0.0;i=2#015#012Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFIzl5c9wbfNG3oaHkl3#015#012From: "15516756486" <sip:15516756486@10.x.x.10>;tag=KXtSBZ1at71Hc#015#012To: <sip:99904441743709685@36.53.xx.xx;transport=udp>;tag=sbcsipuas_1_C35684_20240531083209909_sbc06#015#012Call-ID: a40d965d-99ec-123d-6f92-ecf4bbd64f6c#015#012Contact: <sip:36.53.xx.xx:5060>#015#012CSeq: 84025572 INVITE#015#012Server: sbc_5#015#012Content-Length: 0#015#012#015#012> And the actual SIP message is U 36.53.xx.xx:5060 -> 10.x.x.30:5060 #850 SIP/2.0 607. Via: SIP/2.0/UDP 134.195.195.40:5060 ;branch=z9hG4bKbdbd.95060a558d66b77d258b8d0799be4cf0.0;i=2. Via: SIP/2.0/UDP 127.0.0.8 ;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFCaHXyLA3G.SjwaAmdn. From: "13xxxxxxxxxx" <sip:13xxxxxxxxxx@10.x.x.10>;tag=pFFatBpUc75BH. To: <sip:441xxxxxxxxx@36.53.xx.xx ;transport=udp>;tag=sbcsipuas_1_C56639_20240531081932874_sbc04. Call-ID: e0d2f5fa-99ea-123d-6f92-ecf4bbd64f6c. Contact: <sip:36.53.xx.xx:5060>. CSeq: 84025194 INVITE. Server: sbc_5. Content-Length: 0. Any insights or suggestions would be greatly appreciated. Thank you! Mustafa

5 9

Access avp vars in a cancel? How to pass information from an INVITE to a CANCEL?
by Benoit Panizzon 31 May '24

31 May '24

Hi To determine, if rtpengine was engaged during an invite or not, I make use of rr params for all messages in following transactions and I set an avp for all messages in the current transaction. When the call is canceled, I would need to call rtpengine_delete to make sure rtpengine stops listening for rtp packets. Unfortunately, the avp I set during the INVITE, does not seem to be present while the CANCEL message is processed, despite to my understanding being in the same transaction. How can I pass some information from an INVITE to the corresponding CANCEL? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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