QuincyGao created an issue (kamailio/kamailio#4190)
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### Description
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Explain what you did, what you expected to happen, and what actually happened.
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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
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#### Debugging Data
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```
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```
#### Log Messages
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```
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#### SIP Traffic
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```
(paste your sip traffic here)
```
### Possible Solutions
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### 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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Message ID: <kamailio/kamailio/issues/4190(a)github.com>
al286320 created an issue (kamailio/kamailio#4298)
Hi all,
I'm new to Kamailio, and maybe this is a silly question.
I’ve set up Kamailio as a SIP proxy with two Asterisk servers acting as media services.
This is my configuration for the dispatcher module:
```
loadmodule "dispatcher.so"
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "ds_ping_interval", 20)
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_probing_threshold", 5)
modparam("dispatcher", "ds_inactive_threshold", 5)
modparam("dispatcher", "ds_ping_latency_stats", 1)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3")
modparam("dispatcher", "ds_ping_from", SIPCALL)
```
I can see the SIP OPTIONS requests going from Kamailio to Asterisk and receiving 200 OK responses.

However, the dispatcher is still setting the IP (Inactive Probing) flag on the servers:
```
kamcmd dispatcher.list
{
NRSETS: 1
RECORDS: {
SET: {
ID: 1
TARGETS: {
DEST: {
URI: sip:172.16.254.21:5060
FLAGS: IP
PRIORITY: 0
LATENCY: {
AVG: 0
STD: 0
EST: 0
MAX: 0
TIMEOUT: 183
}
}
DEST: {
URI: sip:172.16.254.20:5060
FLAGS: IP
PRIORITY: 0
LATENCY: {
AVG: 0
STD: 0
EST: 0
MAX: 0
TIMEOUT: 183
}
}
}
}
}
}
```
So when I try to use `ds_select_dst("1", "4")`, it always fails.
Did I miss something in my configuration?
Thanks!
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kerozin created an issue (kamailio/kamailio#4294)
Kamailio 5.7.4
When I write A new value to $fU pseudovariable it acttually appends, but not replace.
A from header before manipulation: <sip:3451088@192.168.107.17>
In failure route I do
$fU = "77719365301";
The header after manipulation looks as: <sip:345108877719365301@192.168.107.17>
this behavior was reported in #3165 but closed. I believe this was happen by mistake.
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Dear Gang
Possibly @oej could provide more in-depth information as he has witnessed this issue.
Usually the user of the from URI is the phone number displayed at the destination. There are situations where this phone number is translated.
As example. In Switzerland, the user is used to see numbers in a local format. National number starting with 0 and international numbers with 00 but on interconnection between telcos, e164 is used.
So basically when a call is sent to a customer '+41' is replaced by '0' and '+' is replaced by '00'.
Let's start with an example From: header:
`From: "Maurice Moss" <sip:+41991234567@example.com>;user=phone`
So shortly before the call is sent out to the location of the registered CPE, this is done:
```
if ($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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therikb31 created an issue (kamailio/kamailio#4296)
For network-initiated UE de-registration, `ims_registrar_scscf` is populating the `Subscription-State` header of the `NOTIFY` message as `active;expires=3581` instead of `terminated`.
### Description
We are simulating a network-initiated UE de-registration by sending a **Registration-Termination Request** from the HSS to the S-CSCF for a registered IMSI.
**Call Flow**:
1. UE registers to the IMS core
2. S-CSCF performs third-party registration with the Application Server on behalf of the UE
3. Application Server subscribes to `Event: reg` for the registered IMSI
4. HSS sends a Registration-Termination Request for network-initiated UE de-registration
5. S-CSCF sends a `NOTIFY` to subscribers for the `IMS_REGISTRAR_CONTACT_DEREGISTERED` event (issue with the `NOTIFY` SIP message)
**Observation**:
1. `NOTIFY` messages are initiated to subscribers of `Event: reg`, but instead of having the `Subscription-State` header set to **terminated**, we observe the value **active;expires=3581**.
(Image for the `NOTIFY` SIP message packet is attached below)

### Troubleshooting
#### Reproduction
To reproduce this, we are performing the following steps:
1. Performing SIP registration of the UE with the target S-CSCF node
2. Sending `SUBSCRIBE` requests from a third-party Application Server and the UE with `Event: reg` to get notified of registration events
3. HSS sends a `Registration-Termination Request` to the target S-CSCF for the registered IMSI (network-initiated de-registration)
#### Debugging Data
#### Log Messages
While going through the logs, we find that this header is being populated based on the expiry timer and a comparison with the current timestamp:
```
2025-06-20 12:45:20 | 19(56) DEBUG: ims_registrar_scscf [registrar_notify.c:1716]: create_notifications(): Expires is greater than current time! Subscription state: [active;expires=3581]
```
#### SIP Traffic
### Possible Solutions
**Workaround**:
In the `create_notifications()` method of `src/modules/ims_registrar_scscf/registrar_notify.c`,
when populating the `Subscription-State` header, if the `event_type` is `10` (`IMS_REGISTRAR_CONTACT_DEREGISTERED`), we can populate `Subscription-State` as `terminated`.
### Additional Information
---
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