September 2023 - sr-dev

[kamailio/kamailio] http_client/http_async_client crash at startup if used with tls and openssl 1.1 (Issue #2912)
by kdanderso 01 Jun '26

01 Jun '26

### Description When attempting to use either http_client or http_async_client with tls support enabled and openssl version 1.1.x, Kamailio crashes at startup. If openssl is reverted to version 1.0.2r as described in #1860, the issue does not occur. #### Reproduction On a system using openssl version 1.1.x (e.g. 1.1.1l) start Kamailio 5.5.2 with this config file for http_client failure: [kamailio.cfg.txt](https://github.com/kamailio/kamailio/files/7442436/kamail… To test http_async_client, load pv.so and http_async_client.so instead of http_client.so. #### Debugging Data [bt.txt](https://github.com/kamailio/kamailio/files/7442493/bt.txt) #### Log Messages [kam.log.txt](https://github.com/kamailio/kamailio/files/7442496/kam.log.txt) ### Additional Information * **Kamailio Version**: [kamversion.txt](https://github.com/kamailio/kamailio/files/7442506/kamversi… * **Operating System**: Linux 5.4.0-89-generic #100-Ubuntu SMP Fri Sep 24 14:50:10 UTC 2021 x86_64 x86_64 x86_64 GNU/Linux [os.txt](https://github.com/kamailio/kamailio/files/7442509/os.txt) -- You are receiving this because you are subscribed to this thread. Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/2912

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[tracker] Assignee added: Assignment operators don't work
by sip-router 29 May '26

29 May '26

THIS IS AN AUTOMATED MESSAGE, DO NOT REPLY. A user has added themself to the list of users assigned to this task. FS#100 - Assignment operators don't work User who did this - Alex Hermann (axlh) http://sip-router.org/tracker/index.php?do=details&task_id=100 You are receiving this message because you have requested it from the Flyspray bugtracking system. If you did not expect this message or don't want to receive mails in future, you can change your notification settings at the URL shown above.

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[kamailio/kamailio] Issue with usrloc 'handle_lost_tcp' and dmq_usrloc 'usrloc_delete' (Issue #3479)
by alternatenetwork 02 May '26

02 May '26

### Description When a TLS connection is closed and handle_lost_tcp=1 the entry is deleted for usrloc. This deletion is not synced via DMQ even though usrloc_delete=1. There is also an issue on the same server if a re-registration happens right after before the timeout was supposed to expire then $var(sv_res) = 2 even though the usrloc table is empty. So it seems like the way handle_lost_tcp deletes is not deleting fully in registrar. usrloc db_mode is 0 ### Troubleshooting #### Reproduction Setup 2 servers with DMQ and close the TCP connection. ### Additional Information * **Kamailio Version** - output of `kamailio -v` ``` version: kamailio 5.6.1 (x86_64/linux) d8f98b flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d8f98b compiled on 19:05:04 Aug 16 2022 with gcc 8.3.0 ``` * **Operating System**:  ``` Debian 10 ``` -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3479 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3479(a)github.com>

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[kamailio/kamailio] Kamailio 5.4.x dialog profiles functionality dead-lock on certain high-load scenarios (Issue #2923)
by Henning Westerholt 03 Apr '26

03 Apr '26

### Description The Kamailio 5.4.x dialog profiles functionality can lead to dead-lock on certain high-load scenarios. The Kamailio dialog profiles are used to track parallel channels for about 200 outgoing PSTN carrier interconnections. During high traffic times (like several thousands parallel calls) the Kamailio server will frequently (e.g. hourly) goes into an end-less loop while executing get_profile_size in the configuration script. This causes the locking for the dialog profiles never be released and Kamailio will stop serving traffic. Internal monitoring tools and RPC commands stay working, as long as they do not touch the dialog functionality. A similar (dedicated) Kamailio setup is used for tracking parallel channels for customers. Here the dead-lock is not observed that frequently, but aparentely also some crashes happens in a much longer time interval. ### Troubleshooting After analysis of the back-traces with GDB the get_profile_size() function was removed from the configuration script. After this change the crash did not happened anymore for several days. #### Reproduction Issue could not be reproduced so far. #### Debugging Data ##### bt 1 (some data removed) (gdb) bt \# 0 0x00007f57cf3b00da in get_profile_size (profile=0x7f50ccbc7e80, value=0x7ffd9928f300) at dlg_profile.c:859 n = 364 i = 12 ph = 0x7f50d3e4b7d0 \# 1 0x00007f57cf419c67 in w_get_profile_size_helper (msg=0x7f57d699d418, profile=0x7f50ccbc7e80, value=0x7ffd9928f300, spd=0x7f57d6916960) at dialog.c:941 \# 2 0x00007f57cf41a459 in w_get_profile_size3 (msg=0x7f57d699d418, profile=0x7f50ccbc7e80, value=0x7f57d6935118, result=0x7f57d6916960) at dialog.c:982 \# 3 0x0000000000463fea in do_action (h=0x7ffd99293610, a=0x7f57d6936488, msg=0x7f57d699d418) at core/action.c:1094 \# 4 0x00000000004711ee in run_actions (h=0x7ffd99293610, a=0x7f57d6936488, msg=0x7f57d699d418) at core/action.c:1581 \# 5 0x000000000046058b in do_action (h=0x7ffd99293610, a=0x7f57d690fda8, msg=0x7f57d699d418) at core/action.c:700 The first back-trace was taking from a running process with gdb. The counter in f0 does not increased that much during this time, probably due the overflow of the loop counter. ##### bt2 (analysis with data structure with gdb scripts) Here the loop counter in f0 showed a really high value. Expected size of dialog profiles hash table: (gdb) p profile->entries[3] $4 = {first = 0x7f9bfd4aad98, content = 2068} (gdb) p profile->entries[7] $3 = {first = 0x7f9c12079f70, content = 784} (gdb) p profile->entries[12] $6 = {first = 0x7f9c02be5d50, content = 7600} (gdb) p profile->entries[14] $2 = {first = 0x7f9bff636de8, content = 6764} hash table bucket 14 shows a lot of corruption and the loop never ends (carrier names and IPs replaced). The list for hash bucket 7 got linked to the list for hash bucket 14: counter 6755: prev 0x7f9c0b9dcde0 - current 0x7f9c02e5b378 - next 0x7f9c0a5f9ba0 - value carrier1-XX.XX - hash 14 counter 6756: prev 0x7f9c02e5b378 - current 0x7f9c0a5f9ba0 - next 0x7f9c0860b968 - value carrier1-XX.XX▒▒▒▒ - hash 14 counter 6757: prev 0x7f9c0a5f9ba0 - current 0x7f9c0860b968 - next 0x7f9bfe3f3a78 - value carrier1-XX.XX▒▒▒▒ - hash 14 counter 6758: prev 0x7f9c0860b968 - current 0x7f9bfe3f3a78 - next 0x7f9c10d977f0 - value carrier1-XX.XX - hash 14 counter 6759: prev 0x7f9bfe3f3a78 - current 0x7f9c10d977f0 - next 0x7f9c0ae198b0 - value carrier2-XX.XX▒▒▒▒ - hash 7 counter 6760: prev 0x7f9c10d977f0 - current 0x7f9c0ae198b0 - next 0x7f9c12079f70 - value carrier3-XX.XX - hash 7 counter 6761: prev 0x7f9c0ae198b0 - current 0x7f9c12079f70 - next 0x7f9c011f2540 - value-carrier2-XX.XX▒▒▒▒ - hash 7 counter 6762: prev 0x7f9c12079f70 - current 0x7f9c011f2540 - next 0x7f9bfff886f0 - value carrier2-XX.XX▒▒▒▒ - hash 7 counter 6763: prev 0x7f9c011f2540 - current 0x7f9bfff886f0 - next 0x7f9c05db00a8 - value carrier3-XX.XX= - hash 7 [...] counter 28270: prev 0x7f9c019d06e8 - current 0x7f9bfaf18290 - next 0x7f9c12c90680 - value carrier2-XX.XX▒▒▒▒ - hash 7 counter 28271: prev 0x7f9bfaf18290 - current 0x7f9c12c90680 - next 0x7f9c086a2b58 - value-carrier2-XX.XX▒▒▒▒ - hash 7 counter 28272: prev 0x7f9c12c90680 - current 0x7f9c086a2b58 - next 0x7f9c0b4f09e8 - value carrier2-XX.XX▒▒▒▒ - hash 7 [...] hash table bucket 7 is still consistent regarding the loop, but already shows initial sign of corruption. There is one item of the list for hash bucket 14 visible: counter 780: prev 0x7f9c0db57ac8 - current 0x7f9c02225700 - next 0x7f9bfbf7db08 - value carrier2-XX.XX▒▒▒▒ - hash 7 counter 781: prev 0x7f9c02225700 - current 0x7f9bfbf7db08 - next 0x7f9c10d977f0 - value carrier1-XX.XX- hash 14 counter 782: prev 0x7f9bfe3f3a78 - current 0x7f9c10d977f0 - next 0x7f9c0ae198b0 - value carrier2-XX.XX▒▒▒▒ - hash 7 counter 783: prev 0x7f9c10d977f0 - current 0x7f9c0ae198b0 - next 0x7f9c12079f70 - value carrier3-XX.XX - hash 7 total size of hash table is 784 #### Log Messages No special log messages observed. #### SIP Traffic SIP traffic looked ok during analysis of the core dumps. ### Possible Solutions * adding additional safe-guards for the get_profile_size function to not access data from other hash buckets * stopping the loop counter after some threshold * finding and fixing the source of the internal data corruption (obviously) * refactoring the dialog modules to use another approach for storing the dialog profile information ### Additional Information * **Kamailio version**: Kamailio 5.4.7, compiled from git repository * **Operating System**: CentOS 7.9 -- You are receiving this because you are subscribed to this thread. Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/2923

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[kamailio/kamailio] siptrace: frong IPv4 address value (#2092)
by sergey-safarov 25 Feb '26

25 Feb '26

### Description Please check attached pcap. [siptrace.pcap.gz](https://github.com/kamailio/kamailio/files/3688791/siptra… Kamailio send INVITE (packet 1) from port `TCP/42394`. SIP trace encapsulate INVITE (packet 2) and encodes Kamailio source port `TCP/5060` #### Debugging Data HEP header parsed by lua [disector](https://github.com/sipcapture/hep-wireshark) ``` HEP3 Protocol HEP ID: HEP3 Length (Bytes): 1574 Protocol family: IPv4 Protocol ID: TCP Source IPv4 address: 172.21.0.131 Destination IPv4 address: 192.168.57.11 Source port: 5060 Destination port: 5060 Timestamp: 1570150645 Timestamp us: 740165 Protocol Type: SIP Capture ID: 1 Payload [truncated]: INVITE sip:911@192.168.57.11;transport=tcp SIP/2.0\r\nRecord-Route: <sip:172.21.0.131;transport=tcp;lr=on;ftag=KvNj7jXrmK89p>\r\nVia: SIP/2.0/TCP 172.21.0.131;branch=z9hG4bK7c2d.2d8104273b7063309f5694ac963c9ad8.0;i=21 ``` [HEP format description located here](https://github.com/sipcapture/HEP/blob/master/docs/HEP3_Network_Proto…. ### Additional Information * **Kamailio Version** - output of `kamailio -v` ``` / # kamailio -v version: kamailio 5.1.6 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 11:01:40 Oct 4 2018 with gcc 6.4.0 ``` * **Operating System**: Docker container Userland ``` / # cat /etc/os-release NAME="Alpine Linux" ID=alpine VERSION_ID=3.8.1 PRETTY_NAME="Alpine Linux v3.8" HOME_URL="http://alpinelinux.org" BUG_REPORT_URL="http://bugs.alpinelinux.org" ``` Kernel Amazon Linux 2 ``` / # uname -a Linux proxy-dc0-0 4.14.128-112.105.amzn2.x86_64 #1 SMP Wed Jun 19 16:53:40 UTC 2019 x86_64 Linux ``` -- You are receiving this because you are subscribed to this thread. Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/2092

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git:master: pv: use suip2a(...) for $snd(..)
by Daniel-Constantin Mierla 04 Dec '25

04 Dec '25

Module: sip-router Branch: master Commit: e151a9e99b1fed7805eaed5d16ca3d6466e1f517 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=e151a9e… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Mon Jul 18 22:37:07 2011 +0200 pv: use suip2a(...) for $snd(..) - fixes FS#140, reported by Klaus Darilion --- modules_k/pv/pv_branch.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/modules_k/pv/pv_branch.c b/modules_k/pv/pv_branch.c index a9dd1bc..4081429 100644 --- a/modules_k/pv/pv_branch.c +++ b/modules_k/pv/pv_branch.c @@ -395,7 +395,7 @@ int pv_get_sndto(struct sip_msg *msg, pv_param_t *param, (int)snd_inf->len); default: /* 0 - ip */ - s.s = su2a(snd_inf->to, sizeof(*snd_inf->to)); + s.s = suip2a(snd_inf->to, sizeof(*snd_inf->to)); s.len = strlen(s.s); return pv_get_strval(msg, param, res, &s); }

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[kamailio/kamailio] misc: examples; ims: Update kamailio.cfg example to function out of the box on current releases (#2203)
by Nick 26 Nov '25

26 Nov '25

Changes to example files for PCSCF in misc/examples/ims to make funcitional with current stable version.  #### Pre-Submission Checklist    - [x] Commit message has the format required by CONTRIBUTING guide - [x] Commits are split per component (core, individual modules, libs, utils, ...) - [x] Each component has a single commit (if not, squash them into one commit) - [x] No commits to README files for modules (changes must be done to docbook files in `doc/` subfolder, the README file is autogenerated) #### Type Of Change - [x] Small bug fix (non-breaking change which fixes an issue) - [ ] New feature (non-breaking change which adds new functionality) - [ ] Breaking change (fix or feature that would change existing functionality) #### Checklist:  - [ ] PR should be backported to stable branches - [x] Tested changes locally - [ ] Related to issue #XXXX (replace XXXX with an open issue number) #### Description Changes to example files for PCSCF in misc/examples/ims to make funcitional with current stable version. Changes: - Loading IPsec module prior to IMS Usrloc PCSCF (Now required) - removed modparam("ims_usrloc_pcscf", "hashing_type", 2) from example (This parameter was removed some time ago) - Fix to formatting of single MySQL connection to work in current version - Bind to any IP by default - Dispatcher parameters only loaded if required You can view, comment on, or merge this pull request online at: https://github.com/kamailio/kamailio/pull/2203 -- Commit Summary -- * misc: examples: IMS PCSCF kamailio.cfg update * misc: examples: IMS PCSCF pcscf.cfg update -- File Changes -- M misc/examples/ims/pcscf/kamailio.cfg (6) M misc/examples/ims/pcscf/pcscf.cfg.sample (12) -- Patch Links -- https://github.com/kamailio/kamailio/pull/2203.patch https://github.com/kamailio/kamailio/pull/2203.diff -- You are receiving this because you are subscribed to this thread. Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/pull/2203

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[kamailio/kamailio] core parsing of SIP message failed (Issue #3062)
by soroshsabz 11 Sep '25

11 Sep '25

ITNOA ### Description I try to connect Kamailio and Freeswitch together, and initiate call from Zoiper5 soft phone through Kamailio My Kamailio config is below ``` #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_DEBUG #!define DBGLEVEL 3 #!else #!define DBGLEVEL 2 #!endif #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!trydef DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */ log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases - it can be set many times */ # alias="sip.mydomain.com" /* listen sockets - if none set, Kamailio binds to all local IP addresses * - basic prototype (full prototype can be found in Wiki - Core Cookbook): * listen=[proto]:[localip]:[lport] advertise [publicip]:[pport] * - it can be set many times to add more sockets to listen to */ listen=udp:172.28.72.99:5060 advertise 172.28.72.99:5060 #listen=tcp:172.28.72.99:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_JSONRPC tcp_accept_no_cl=yes #!endif #!ifdef WITH_TLS enable_tls=no /* upper limit for TLS connections */ tls_max_connections=2048 #!endif /* set it to yes to enable sctp and load sctp.so module */ enable_sctp=no ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* 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 = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_JSONRPC loadmodule "xhttp.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "textopsx.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) /* set it to 0 to disable Path handling */ modparam("registrar", "use_path", 1) /* save Path even if not listed in Supported header */ modparam("registrar", "path_mode", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- modparam("usrloc", "timer_interval", 60) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks # route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); # handle requests within SIP dialogs # route(WITHINDLG); ### only initial requests (no To tag) # authentication # route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers #remove_hf("Route"); #if (is_method("INVITE|SUBSCRIBE")) { # record_route(); #} # account only INVITEs #if (is_method("INVITE")) { # setflag(FLT_ACC); # do accounting #} # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests #route(PRESENCE); # handle registrations # route(REGISTRAR); #if ($rU==$null) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; #} # dispatch destinations to PSTN #route(PSTN); # user location service #route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly|scanner|sipcli|sipvicious|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("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing 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") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE #!ifdef WITH_MSGREBUILD # apply changes in case the request headers or body were modified msg_apply_changes(); #!endif if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif 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(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-Hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # JSONRPC over HTTP(S) routing #!ifdef WITH_JSONRPC event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ``` As you can see I just one thing in `request_route` ``` request_route { # per request initial checks # route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); ``` The `172.28.69.52` is ip of my Freeswitch After I try to call from Zoiper5, I get below error in Kamailio log and Zoiper does not work. ``` 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) ``` ### Troubleshooting #### Reproduction * pull official Kamailio docker image `kamailio/kamailio 5.4.6-buster` * then run below command ``` docker run --net=host --name kamailio \ -v /home/ssoroosh/kamailio:/etc/kamailio \ kamailio/kamailio:5.4.6-buster ``` #### Debugging Data I do not have any debug data #### Log Messages ``` Listening on udp: 172.28.72.99 [172.28.72.99]:5060 advertise 172.28.72.99:5060 Aliases: 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:2919]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 13(19) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/19 15(21) INFO: ctl [io_listener.c:210]: io_listen_loop(): io_listen_loop: using epoll_lt as the io watch method (auto detected) 2(8) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<�> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<�> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) ``` #### SIP Traffic My pcap is attached [sip.zip](https://github.com/kamailio/kamailio/files/8326814/sip.zip) ### Possible Solutions I have not any solution yet. ### Additional Information * **Kamailio Version** - output of `kamailio -v` ``` version: kamailio 5.4.6 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 ``` * **Operating System**: ``` Linux master 5.4.0-104-generic #118-Ubuntu SMP Wed Mar 2 19:02:41 UTC 2022 x86_64 GNU/Linux root@master:/home# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster ``` -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3062 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3062(a)github.com>

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[kamailio/kamailio] Writing to $fU or $tU appends to header user (Issue #3165)
by Benoît Panizzon 24 Jun '25

24 Jun '25

Dear Gang Possibly @oej could provide more in-depth information as he has witnessed this issue. Usually the user of the from URI is the phone number displayed at the destination. There are situations where this phone number is translated. As example. In Switzerland, the user is used to see numbers in a local format. National number starting with 0 and international numbers with 00 but on interconnection between telcos, e164 is used. So basically when a call is sent to a customer '+41' is replaced by '0' and '+' is replaced by '00'. Let's start with an example From: header: `From: "Maurice Moss" <sip:+41991234567@example.com>;user=phone` So shortly before the call is sent out to the location of the registered CPE, this is done: ``` if ($fU =~ "^\+41") { $fU = "0" + $(fU{s.substr,3,0}); } else if ($fU = ~ "^\+") { $fU = "00" + $(fU{s.substr,1,0}); } ``` What is sent to the CPE now looks like this: `From: "Maurice Moss" <sip:0991234567@example.com>;user=phone` Now we hit an error like 486 BUSY and the destination has call forwarding active to a mobile phone on another TSP. So we have to send the call out back the IC and numbers need to be translated back to e164. We handle this in a failure route, which in turn could trigger a branch route. So we revert the number back to e164: `$fU = "+41" + $(fU{s.substr,1,0});` Expected outcome: `From: "Maurice Moss" <sip:+41991234567@example.com>;user=phone` Observed outcome: `From: "Maurice Moss" <sip:0991234567+41991234567@example.com>;user=phone` So setting $fU more than once is appending to the user element of the From header URI. This behavior has not been found in any documentation. I have been working around most of the issues by making sure I change $fU (and $tU) at the latest possible time and only once. But in the case described above, I have not been able to come up with a work-around yet. I also can't think of any benefit of the way those PV are handled or any harm that could be done, to handle them differently and make the last 'write' overwrite and previous value, instead of appending. Thank you for looking into this. -Benoît- -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3165 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3165(a)github.com>

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[kamailio/kamailio] Inactive video stream: can't find media IP in the message (Issue #3406)
by schoberw 31 May '25

31 May '25

### Description In some situations the other side UA must reply with an inactive media stream. E.g. if video was added but not supported by the other side. The UAS then must answer with an m=video 0 RTP/.. line. It seams that the video section must contain a c= line with an IP address for rtpengine to function. If there is no c line (no IP address) the SDP body cannot be parsed and thus no RTP proxy is invoked. ### Troubleshooting Not easy to reproduce since the answer must bei "wrong". Is it correct to reply an SDP body (media = video) like this? ``` m=video 0 RTP/AVPF 96 a=label:1 a=inactive a=mid:1 ``` Maybe not (but Audiocodes SBC lates LTS version does it) - so we are dependent on other side now that this service works. #### Reproduction reInvite with Video to a UA that has no video Support (e.g. Bria -> Snom). Drop the c line in the video part with some textops in 200 OK (just to reproduce of course). A -> B, connect the call A -> + video, B has no video support A presses hold #### Log Messages Incoming 200 OK after doing hold with an inactive video: ``` v=0 o=root 436040161 309284577 IN IP4 1.2.3.5 s=call t=0 0 m=audio 14200 RTP/AVPF 9 8 126 c=IN IP4 1.2.3.5 a=ptime:20 a=recvonly a=mid:0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-15 m=video 0 RTP/AVPF 96 a=label:1 a=inactive a=mid:1 Mar 27 17:52:22 c5p05es2-1 /usr/sbin/kamailio[2938205]: ERROR: <core> [core/parser/sdp/sdp.c:490]: parse_sdp_session(): can't find media IP in the message Mar 27 17:52:22 c5p05es2-1 /usr/sbin/kamailio[2938205]: INFO: <script>: >>> Sending Reply: 200 OK (1.2.3.4:443 -> 100.108.48.38:65251) ``` Compare to working SDP parsing: ``` a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-15 m=video 0 RTP/AVP 96 c=IN IP4 1.2.3.5 a=inactive a=mid:1 Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: ONREPLY_ROUTE[FORWARD] Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: ROUTE[RTPP_REPLY] Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > 200 with video Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > 200 with inactive video Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > Remove inactive video Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > Answer to WebRTC client: 200 - qijpuibkv6ufhnd282ks Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: >>> Sending Reply: 200 OK (91.237.65.14:443 -> 80.108.48.38:65388) ``` ### Possible Solutions Tried to fix in script: ``` if(sdp_with_media("video")) xlog("L_INFO", "> $rs with video"); if(sdp_with_active_media("video")) xlog("L_INFO", "> $rs with active video"); if(!sdp_with_active_media("video")) xlog("L_INFO", "> $rs with inactive video"); if(sdp_with_media("video") && !sdp_with_active_media("video")) { # try fix for RMT#60189 xlog("L_INFO", "> Remove inactive video"); sdp_remove_media("video"); msg_apply_changes(); # needed? } ``` But since sdpops cannot parse it cannot remove the buggy media, too. See the log: non of the xlog is logging. If there is inactive media the c line is not mandatory. The parser should accept this by adding a default IP of 0.0.0.0, if needed Doing tricks with textops is not a easy nor performant solution to accept this wrong SDP from other parties. ### Additional Information ``` version: kamailio 5.6.4 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 ``` * **Operating System**: Debian Bullseye -- Reply to this email directly or view it on GitHub: https://github.com/kamailio/kamailio/issues/3406 You are receiving this because you are subscribed to this thread. Message ID: <kamailio/kamailio/issues/3406(a)github.com>

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