longchun4835 created an issue (kamailio/kamailio#4178)
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### Description
Using function sip_trace_mode() with parameter d to enable dialog tracing mode logs:
```
BUG: siptrace [siptrace.c:1671]: add_info_xavp(): URI should be in raw format here
ERROR: siptrace [siptrace.c:888]: sip_trace_helper(): failed to serialize siptrace info! Won't trace dialog!
```
### Troubleshooting
#### Reproduction
my module parameters:
```
#!ifdef WITH_HOMER
loadmodule "siptrace.so"
modparam("siptrace", "duplicate_uri", HOMER_URL)
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "hep_version", 3)
modparam("siptrace", "hep_capture_id", HOMER_HEP_ID)
#!endif
...
request_route {
...
if (is_method("INVITE")) {
sip_trace_mode("d");
}
}
```
### Possible Solutions
I checked the source code and found that the issue occurs when sip_trace_helper() is called inside the ki_sip_trace_mode() function, with the third parameter (duri) set to NULL. I don’t quite understand the logic behind passing NULL does this mean that the 'D' parameter will never working when calling sip_trace_mode()?
Additionally, I have another question: in the sip_trace_helper() function, why does it call trace_transaction() instead of trace_dialog() when strace_type == SIPTRACE_DIALOG?
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
version: kamailio 5.6.4 (x86_64/linux) a004cf
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
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: a004cf
compiled on 09:14:30 Jan 23 2025 with gcc 8.3.1
```
* **Operating System**:
```
NAME="CentOS Linux"
VERSION="7 (Core)"
ID="centos"
ID_LIKE="rhel fedora"
VERSION_ID="7"
PRETTY_NAME="CentOS Linux 7 (Core)"
ANSI_COLOR="0;31"
CPE_NAME="cpe:/o:centos:centos:7"
HOME_URL="https://www.centos.org/"
BUG_REPORT_URL="https://bugs.centos.org/"
```
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sergey-safarov created an issue (kamailio/kamailio#4241)
### Description
In the GitHub Actions present Alpine Docker image build error
```
tar (child): kamailio_min-without_os_files.tar.gz: Cannot open: No such file or directory
tar (child): Error is not recoverable: exiting now
tar: Child returned status 2
tar: Error is not recoverable: exiting now
```
https://github.com/kamailio/kamailio/actions/runs/14722992691/job/413201110…
With high probability, this does not work
I will work to fix this, but if possible, it will be fine to switch to use `cmake` for Alpine Docker image.
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IgorrG created an issue (kamailio/kamailio#4268)
### Description
We use RPC dlg.end_dlg command to terminate dialog. In most cases it does operate properly and send BYE from kamailio to both sides of dialog (as show on screenshot).

We found that in some cases we have only first BYE sent to calling party. Second BYE never sent, when called party tries to terminate call by timeout it get responded with 481.


In logs we have such warning:
мая 27 15:11:46 service-proxy.iqtek.ru kamailio[3518388]: 2(3518388) WARNING: {1 1 BYE 541d3f2c-43fd-43b6-af24-bad1fbf85eb6} dialog [dlg_handlers.c:1343]: dlg_onroute(): unable to find dialog for BYE with route param 'e1e.4f22' [3614:8948] and call-id '541d3f2c-43fd-43b6-af24-bad1fbf85eb6'
We also found that in this case this warning shown in logs 0.04 seconds after kamailio receive 200ok.
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
# ./kamailio -V
version: kamailio 5.8.6 (x86_64/linux) fb71db
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_SEND_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: fb71db
compiled on 17:38:27 May 26 2025 with gcc 12.2.0
```
* **Operating System**:
```
# lsb_release -a
No LSB modules are available.
Distributor ID: Debian
Description: Debian GNU/Linux 12 (bookworm)
Release: 12
Codename: bookworm
# uname -a
Linux rtp-kamailio2 6.1.0-32-amd64 #1 SMP PREEMPT_DYNAMIC Debian 6.1.129-1 (2025-03-06) x86_64 GNU/Linux
```
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IgorrG created an issue (kamailio/kamailio#4266)
### Description
While implementing mid-register inspired by pull request (https://github.com/kamailio/kamailio/pull/3360) we found the case when it goes impossible to handle responses for registrar servers. In case some of registrars busy and respond with 401 after 200ok received - it is not possible using any routes and configuration options to handle responses received in branch transactions.
According to documentation I think that setting wt_timer should allow to absorb and handle responses on failure-route while transaction kept in memory. In fact wt_timer does not affect handling responses in any way.

### Troubleshooting
To reproduce this case we have configured sipp to response instant with 200ok on registration request.
#### Reproduction
To reproduce issue SIPP could be setup as one of the dispatcher nodes with instant 200ok reply to register. Kamailio should use following configuration snippet: https://github.com/ovoshlook/kamailio-mid-registrar-config-snippets/blob/ma…
#### Log Messages
REGISTER|401tm [t_lookup.c:912]: t_reply_matching(): t_reply_matching: hash 12337 label 0 branch 1
REGISTER|401tm [t_lookup.c:986]: t_reply_matching(): reply (0x7f1fc67f83a8) matched an active transaction (T=0x7f1fc0f0f740)!
REGISTER|401tm [t_lookup.c:1122]: t_check_msg(): msg (0x7f1fc67f83a8) id=13/1915035 global id=13/1915035 T end=0x7f1fc0f0f740
REGISTER|401tm [t_reply.c:2363]: reply_received(): transaction found - T:0x7f1fc0f0f740 branch:1
REGISTER|401tm [t_reply.c:2376]: reply_received(): original status uas=200, uac[1]=0 local=0 is_invite=0)
REGISTER|401tm [t_reply.c:1363]: t_should_relay_response(): ->>>>>>>>> T_code=200, new_code=401
REGISTER|401tm [t_reply.c:1374]: t_should_relay_response(): final reply already sent
REGISTER|401tm [t_reply.c:1625]: t_should_relay_response(): finished with rps discarded - uas status: 200
### Possible Solutions
<!--
If you found a solution or workaround for the issue, describe it. Ideally, provide a pull request with a fix.
-->
### Additional Information
* **Kamailio Version**
```
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 on 16:35:08 Jun 5 2023 with gcc 10.2.1
```
* **Operating System**:
```
No LSB modules are available.
Distributor ID: Debian
Description: Debian GNU/Linux 11 (bullseye)
Release: 11
Codename: bullseye
Linux pbxKamaTest 5.10.0-23-amd64 #1 SMP Debian 5.10.179-1 (2023-05-12) x86_64 GNU/Linux
```
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### Description
After carrying out some load tests using the `Websocket` module, I noticed that sometimes the `websocket:closed` event is not triggered.
This problem seems to be particularly noticeable when the load is high and simultaneous. Example: 1000 records simultaneously.
In the tests carried out, I realized that the number of times the event is not triggered is small, in the order of [1-10]/1000.
I check that when I do a `ws.dump` I don't have any active connections, and I confirm that they are all effectively cleaned (`netstat -ano | grep 8061`).
However, the event is not triggered for some connections.
I can see this in the logs, and also when i display the content of an hash table, i'm dropping key inside `websocket:closed` event.
### Troubleshooting
The troubleshooting was simplistic, i.e. I ran the same test n times, and validated that the event was triggered for all connections, the number of `ws` connections listed by `Kamailio`, and the open connections.
#### Reproduction
To reproduce, we can use any mechanism that emulates the use of the `Websocket` module.
For example, I used the `sipexer` tool to create a `wss` connection with `Kamailio`, forcing simultaneous registrations.
Example command: `sipexer -cb -mt register -ex 60 -au <user> -ap <pass> -fuser example -fd mydomain -wso https://10.0.0.12:10000 -ruri sip:kamailio -su wss://kamailio:8061`
Kamailio TLS listener:
`listen=tls:10.0.0.12:8061 advertise PUBLIC_IP`
Websocket module:
```
loadmodule "websocket.so"
tcp_accept_no_cl=yes
```
```
event_route[websocket:closed] {
xlog("L_NOTICE", "WebSocket connection closed $proto:$si:$sp\n");
<I also delete where a htable key>
}
```
#### Debugging Data
```
[root@ ~]$ prcmd ws.dump
{
connections: {
}
info: {
wscounter: 0
truncated: no
}
}
[root@ ~]$ netstat -ano | grep 8061
tcp 0 0 10.0.0.12:8061 0.0.0.0:* LISTEN off (0.00/0/0)
[root ~]$ prcmd htable.dump wsauth | grep "name" | wc -l
3
[root ~]$ cat /var/log/kamailio.log | grep "WebSocket connection closed" | wc -l
997
[root ~]$ cat /var/log/proxy-registrar/proxy-registrar.log | grep "ERR" | grep "WAR" | wc -l
0
```
#### Log Messages
Nothing relevant.
#### SIP Traffic
Nothing relevant here, the SIP Flow is OK.
### Additional Information
```
version: kamailio 5.6.4 (x86_64/linux) a004cf-dirty
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES
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: a004cf -dirty
compiled on 23:17:27 Jun 7 2024 with gcc 4.8.5
```
* **Operating System**:
```
CentOS Linux release 7.9.2009 (Core)
```
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QuincyGao created an issue (kamailio/kamailio#4190)
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### Description
<!--
Explain what you did, what you expected to happen, and what actually happened.
-->
I want to use opensips as SBC, When the agent registers to kamailio, the contact content is modified to the ip and port of kamailio, and then Register is forwarded to the agent.
remove_hf("Contact") can't remove the whole Contact( field, vlue).
here is my config:
```
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
}
fix_nated_contact();
if (!save("location","0x02")) {
#if (!save("location")) {
sl_reply_error();
}
$fs="udp:172.16.4.111:5461";
$du="sip:172.16.4.114:5060";
remove_hf("Contact");
$var(contact) = "Contact: <sip:" +$fU+"@"+"172.16.4.111:5461>;expires=120\r\n" ;
append_hf("$var(contact)");
route(RELAY);
exit;
}
```
here is the sip message:
```
2025/03/26 16:36:07.706093 172.16.4.111:5461 -> 172.16.4.114:5060
REGISTER sip:172.16.4.111:5460 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.111:5461;branch=z9hG4bKfc7e.d2df36f3b5681afeb71e442e645de38d.0
Via: SIP/2.0/UDP 172.16.80.3:49309;received=172.16.80.3;rport=49309;branch=z9hG4bKPj8ab8bd1d3c3a4c1bb0cd639540fa3183
Max-Forwards: 69
From: <sip:1008@172.16.4.111>;tag=95f7f5601f8b40458a1efe6923ab0a29
To: <sip:1008@172.16.4.111>
Call-ID: f84b643f5d284d18a0d8175e1ef4d747
CSeq: 33264 REGISTER
User-Agent: MicroSIP/3.21.6
sip:1008@172.16.80.3:49309;obExpires: 120
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
Contact: <sip:1008@172.16.4.111:5461>;expires=120
```
you can see the problem is that: `sip:1008@172.16.80.3:49309;obExpires: 120` ,
`sip:1008@172.16.80.3:49309` is the value of the Contact field after fix_nated_contact().
### Troubleshooting
#### Reproduction
<!--
If the issue can be reproduced, describe how it can be done.
-->
#### Debugging Data
<!--
If you got a core dump, use gdb to extract troubleshooting data - full backtrace,
local variables and the list of the code at the issue location.
gdb /path/to/kamailio /path/to/corefile
bt full
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list
If you are familiar with gdb, feel free to attach more of what you consider to
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```
(paste your debugging data here)
```
#### Log Messages
<!--
Check the syslog file and if there are relevant log messages printed by Kamailio, add them next, or attach to issue, or provide a link to download them (e.g., to a pastebin site).
-->
```
(paste your log messages here)
```
#### SIP Traffic
<!--
If the issue is exposed by processing specific SIP messages, grab them with ngrep or save in a pcap file, then add them next, or attach to issue, or provide a link to download them (e.g., to a pastebin site).
-->
```
(paste your sip traffic here)
```
### Possible Solutions
<!--
If you found a solution or workaround for the issue, describe it. Ideally, provide a pull request with a fix.
-->
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
version: kamailio 5.8.5 (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, MEM_JOIN_FREE, 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_SEND_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:
compiled on 11:27:01 Mar 21 2025 with gcc 4.8.5
```
* **Operating System**:
<!--
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```
Centos7.9
```
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<!-- All points should be verified, otherwise, read the CONTRIBUTING guidelines from above-->
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in `doc/` subfolder, the README file is autogenerated)
#### Type Of Change
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<!-- Go over all points below, and after creating the PR, tick the checkboxes that apply -->
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- [x] Tested changes locally
- [ ] Related to issue #XXXX (replace XXXX with an open issue number)
#### Description
<!-- Describe your changes in detail -->
In this PR we introduce a new mode and two new parameters for accomplishing the same thing for user part also in domain part.
Docs incoming if no significant changes are needed.
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/4246
-- Commit Summary --
* topos: Add new contact_mode=3 and 2 new modparam for contact host domains
* topos: Refactor contact_mode handling
* topos: Append port and protocol param
-- File Changes --
M src/modules/topos/topos_mod.c (13)
M src/modules/topos/tps_storage.c (428)
M src/modules/topos/tps_storage.h (2)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/4246.patchhttps://github.com/kamailio/kamailio/pull/4246.diff
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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-
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