Having (optional) support for socket activation in Kamailio would bring several benefits:
* Security: systemd can listen on privileged ports or socket paths and start Kamailio with privileges already dropped.
* Major upgrades or reconfigurations of Kamailio requiring a full restart are possible without having the listening socket(s) ever go away.
* Any service dependent on Kamailio doesn't have to wait for it to finish starting (or start at all).
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A sip proxy may serve many domains listed in its SSL certificate X509v3 Subject Alternative Name field. In order to find out if a peer sip proxy is serving a particular domain, access would be needed in config file to this list of domains.
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### Description
While implementing an edgeproxy based on kamailio i have to add a custom SIPHeader which contains the uid from the useragents client certificate.
The UID is part of the subject like:
O = Company
CN = Sebastian Denz
UID = denz
But i cant access that attribute. All i can do is reading the CN and O.
### Expected behavior
There should be a select to access the uid Attribute from the subject, or at least a select which returns the whole subject, so it can be parsed manually in kamailio.cfg.
#### Actual observed behavior
It is only possible to access the O and CN field from the subject.
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
kamailio -v
version: kamailio 5.2.0 (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 with gcc 5.3.1
```
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Hello all,
I've come across the following issue with topos on Kamailio 5.1.0-rc2.
### Description
Kamailio acts as a proxy for asterisk instances residing in a private IP range. Asterisk's peers are configured with their public IP address or hostname, but asterisk will dispatch requests to them to kamailio instead (because kamailio's IP is set in outboundproxy option in peer definition).
This works fine for INVITEs, but if qualify=yes is set for the peer, asterisk will send out OPTIONs to monitor these peers. If the topos module is loaded, 200 OK replies to these OPTIONs will be malformed when forwarded from kamailio to asterisk (no Via header present). Disabling the topos module will resolve this.
### Troubleshooting
#### Reproduction
Configure a peer in asterisk with outboundproxy option set to kamailio's IP address. Set qualify=yes.
Adjust kamailio's default configuration so that OPTIONs to other hosts besides itself are routed. Change the following section in request_route:
```
if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
```
to:
```
if ( ($rU==$null) && !(is_method("OPTIONS")) ) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# Permit asterisk's monitoring using options
if(is_method("OPTIONS") && uri!=myself) {
route(RELAY);
}
```
Enable and configure the topos module with database.
#### SIP Traffic
asterisk:5060 -> kamailio:5060
```
OPTIONS sip:peer.domain.tld SIP/2.0
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK460330cf;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@domain.tld>;tag=as690c2406
To: <sip:peer.domain.tld>
Contact: <sip:asterisk@asterisk:5060>
Call-ID: 6548f43b2d8740122186bb0c6e72ac78(a)domain.tld
CSeq: 102 OPTIONS
User-Agent: myUA
Date: Sun, 03 Dec 2017 00:39:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
```
kamailio:5060 -> peer:5060
```
OPTIONS sip:peer.domain.tld SIP/2.0
Via: SIP/2.0/UDP kamailio:5060;branch=z9hG4bK460330cf;rport
Max-Forwards: 69
From: "asterisk" <sip:asterisk@domain.tld>;tag=as690c2406
To: <sip:peer.domain.tld>
Contact: <sip:asterisk@asterisk:5060>
Call-ID: 6548f43b2d8740122186bb0c6e72ac78(a)domain.tld
CSeq: 102 OPTIONS
User-Agent: myUA
Date: Sun, 03 Dec 2017 00:39:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
```
peer:5060 -> kamailio:5060
```
SIP/2.0 200 OK
Via: SIP/2.0/UDP kamailio;branch=z9hG4bKda8d.cc03bd2def8ad36e8c43f61ab31a636f.0;received=kamailio;rport=5060
From: "asterisk" <sip:asterisk@domain.tld>;tag=as690c2406
To: <sip:peer.domain.tld>;tag=as472874ee
Call-ID: 6548f43b2d8740122186bb0c6e72ac78(a)domain.tld
CSeq: 102 OPTIONS
Server: myUA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:peer:5060>
Accept: application/sdp
Content-Length: 0
```
**kamailio:5060 -> asterisk:5060: Missing Via Header**
```
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@domain.tld>;tag=as690c2406
To: <sip:peer.domain.tld>;tag=as472874ee
Call-ID: 6548f43b2d8740122186bb0c6e72ac78(a)domain.tld
CSeq: 102 OPTIONS
Server: myUA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:peer:5060>
Accept: application/sdp
Content-Length: 0
```
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
version: kamailio 5.1.0-rc2 (x86_64/linux)
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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_LISTEN 16, 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 6.3.0
```
* **Operating System**:
```
Debian 9.1
```
* **Other pertinent configuration information**:
This kamailio box has several network interfaces. mhomed is not used, instead force_send_socket is called where necessary. I'm mentioning this in case it's what might trigger this issue with topos.
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The db_redis module seems to only support connecting with a single redis instance. It is not evident from the documentation if it implements support for native redis cluster by honouring 'MOVED' replies.
ndb_redis seems to have support for native redis clusters, and supports both automatic (see allow_dynamic_nodes parameter of ndb_redis) and manual configuration of redis nodes.
Since both modules are based on the hiredis library, is it safe to assume this support could be easily implemented, with some code sharing between the two modules perhaps?
If db_redis supports clusters with automatic configuration only (by processing MOVED replies), perhaps the module's documentation should be updated to reflect that.
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### Description
Kamailio server has two legs that are connected to different networks.
I'm using Kamailio v.5.2.3 and the "enable_double_rr" is implicitly set to "1".
The leg "A" IP address is 10.28.80.1
The leg "B" IP address is 192.168.9.103
The call is initiated from 10.28.80.47
According to the "rr" module documentation, function record_route()
should insert two "Record_Route" header fields when a request is
accepted on one leg is should go out via the second leg. This works as
expected in case of UDP protocol:
INVITE sip:2601@192.168.0.178:49162;transport=udp SIP/2.0
Record-Route: <sip:192.168.9.103;r2=on;lr;did=e2c.a191>
Record-Route: <sip:10.28.80.1;r2=on;lr;did=e2c.a191>
Via: SIP/2.0/UDP 192.168.9.103;branch=z9hG4bKcfa5.d64ecbd87d5315b5993c4ccf16f86537.0
Via: SIP/2.0/UDP 10.28.80.47:5060;rport=5060;branch=z9hG4bK3a9e9a4d
But when the TCP protocol is used AND "listen" parameters are defined:
listen=udp:10.28.80.1:5060
listen=tcp:10.28.80.1:5060
listen=udp:192.168.9.103:5060
listen=tcp:192.168.9.103:5060
then the outbound message looks like this:
INVITE sip:2601@192.168.0.178:48758;transport=tcp SIP/2.0
Record-Route: <sip:10.28.80.1;transport=tcp;lr;did=bcd.72f>
Via: SIP/2.0/TCP 10.28.80.1;branch=z9hG4bKb9ca.26ea08654c9dbc32bb0dc6e3b6f92d45.0;i=3
Via: SIP/2.0/TCP 10.28.80.47:5060;rport=33976;branch=z9hG4bK5291fab1
There are two problems there:
a) only one Record-Route with leg "A" is inserted
b) the added "Via" header field contains the leg "A" IP address instead of expected leg "B" IP address (192.168.9.103). In the LAN trace, I see that in reality the message was sent from leg "B".
IMPORTANT: The problem occurs only when the "listen" parameters are defined in the Kamailio configuration. When the "listens" are not configured then everything works correctly.
### Troubleshooting
#### Reproduction
#### Debugging Data
#### Log Messages
http://updates.xorcom.com/~xorcom/kam-tcp-problem.tar.gz
The archive contains the following files:
* syslog-bad.log and bad.cap - Kamailio log and tcpdump trace of a call when the problem occurs
* syslog-good.log and good.cap - Kamailio log and tcpdump trace of a call when the problem doesn't occur
#### SIP Traffic
### Possible Solutions
### Additional Information
kamailio -v
version: kamailio 5.2.3 (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 with gcc 7.3.0
***Operating System**:
Ubuntu 18.0.4
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(from devmeeting #Fosdem) - adding new dependencies as core dependencies
- Curl
- libssl
- uuid
- libunistring
- libxml2
Any other candidates? They have to be libraries that are commonly installed on most Linux/Unix system.
This would mean that a lot of more modules will be built by default, like TLS, HTTP_CLIENT, WEBSOCKET.
We may also look over the names of the packages/groups for the build system and linux distros
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### Description
Kamailio seems to fail to properly route in-dialog SUBSCRIBE with the topos module enabled. In this configuration, SUBSCRIBEs are expected to be delivered to a downstream UAS, an asterisk instance in the internal network.
record_route() is executed for all initial requests, including SUBSCRIBEs. Routing is performed correctly with the topos module disabled, no other modifications were needed.
<!--
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### Troubleshooting
I haven't been able to work around this issue in any other way other than disabling the topos module.
#### Reproduction
The configuration used is a heavily modified version of the example configuration file, but the handling of SUBSCRIBEs is generally simple. The initial SUBSCRIBE is subject to some tests, the R-URI is edited and dispatched to a farm of asterisk servers based on some criteria. Record Routing is enforced, so subsequent in-dialog re-SUBSCRIBEs are expected to be routed with loose_route(), which doesn't seem to handle things properly.
<!--
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#### Debugging Data
I'm attaching kamailio logs with debug=4, cfg_trace
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#### Log Messages
See attachments
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#### SIP Traffic
Please see: https://www.cloudshark.org/captures/d2643aae7f2a
<!--
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### Possible Solutions
Using topoh instead for topos, but unfortunately this will break communication for other reasons (messages too large/UDP fragmentation for some peers).
### Additional Information
* **Kamailio Version**
Tried with kamailio 5.1.6 and kamailio 5.3.0-dev0
deb.kamailio.org repositories were used to acquire binaries for both versions
* **Operating System**:
Debian Stretch
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### Description
While debugging an issue with locally generated in-dialog requests I stumbled upon some weirdness with the alias functionality from the nathelper module.
The dialog modules tries to handle an alias parameter when sending an in-dialog request by calling ```uri_restore_rcv_alias()``` in the function ```build_dlg_t()```. A few instructions later, the same is attempted by the tm module in the function ```req_within()```.
It seems to me that the attempt by the dialog module is redundant and can be removed.
And while at it, IMHO it should be removed from the tm module too. This is functionality introduced by the nathelper module. Nowhere in the documentation it is mentioned there is some automatic handling of the alias parameter. If handling is needed for local requests, the script writer can call ```handle_uri_alias()``` from within local_route.
If the functionality has to stay, may I suggest that the code in the nathelper module makes use of the ```uri_restore_rcv_alias()``` function instead of having its own duplicate implementation. Less maintenance, easier debugging.
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### Description
rightnow, there is no stats about command sent to rtpengine and getting reply from rtpengine. There should be a stats that will tell that
number of offer request sent:
number of answer request sent:
number of delete request sent:
number of failed request for offer [after retry]:
number of failed request for answer [after retry]:
number of failed request for delete [after retry]:
This will help to identify actual number of failed request.
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### Troubleshooting
#### Reproduction
<!--
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#### Debugging Data
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```
(paste your debugging data here)
```
#### Log Messages
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```
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#### SIP Traffic
<!--
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-->
```
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```
### Possible Solutions
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### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
(paste your output here)
```
* **Operating System**:
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```
(paste your output here)
```
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