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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
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Actual implementation doesn't work with async. Need to investigate how to support it
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>From https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=886111
```
Source: kamailio
Version: 5.1.0-1
Severity: wishlist
Hi,
I noticed from the build log on mips64el a lot of warnings like this:
> In file included from ../../core/parser/../mem/../atomic/atomic_native.h:50:0,
> from ../../core/parser/../mem/../futexlock.h:42,
> from ../../core/parser/../mem/../lock_ops.h:75,
> from ../../core/parser/../mem/shm.h:39,
> from ../../core/parser/../mem/shm_mem.h:34,
> from ../../core/parser/../ut.h:45,
> from ../../core/parser/../ip_addr.h:39,
> from ../../core/parser/msg_parser.h:37,
> from app_sqlang_api.h:28,
> from app_sqlang_kemi_export.c:32:
> ../../core/parser/../mem/../atomic/atomic_mips2.h:41:2: warning: #warning mips64 atomic code was not tested, please report problems to serdev(a)iptel.org or andrei(a)iptel.org [-Wcpp]
> #warning mips64 atomic code was not tested, please report problems to \
> ^~~~~~~
If possible, kamailio should use the standard c atomic code from
stdatomic.h instead of providing assembly routines. This will improve
maintainability and architecture support.
Thanks,
James
```
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### Description
When using the t_reuse_branch() function after a branch failure, Kamailio doesn't send out the call again if $du was filled and differed from $ru. Instead, Kamailio tries to send the call directly to $ru.
I would expect that the call gets sent out the same way as the original call.
### Troubleshooting
#### Reproduction
You need a setup like this:
UAC1 K(LB) <=> K(Proxy/Registrar) <=> K(LB) => UAC2
When a UAC registers, the Path module is used to save the LB (Loadbalancer) for outbound calls. When UAC1 calls UAC2, the original call gets sent through K(LB) to UAC2. However, if the first call gets rejected and the call gets retried through branch failure route, K(Proxy/Registrar) tries to send the call directly to the request URI.
In your branch_route add a reference to a branch failure route:
```t_on_branch_failure("samplename");```
Then add a branch failure route like this:
```event_route[tm:branch-failure:handle_tls_branch] {
if(t_check_status("488")) {
if (isbflagset(0) && !isbflagset(1)) {
t_reuse_branch();
setbflag(1);
t_relay();
}
}
}```
When filling $du manually before sending out the call again, everything works.
### Possible Solutions
The original destination URI should be copied to the reused branch.
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
version: kamailio 4.4.5 (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 4.7.2
```
But from what I see in source code, the devel version still copies the same value.
* **Operating System**:
<!--
Details about the operating system, the type: Linux (e.g.,: Debian 8.4, Ubuntu 16.04, CentOS 7.1, ...), MacOS, xBSD, Solaris, ...;
Kernel details (output of `uname -a`)
-->
```
Debian Wheezy
```
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Regarding the issue 1274 (https://github.com/kamailio/kamailio/issues/1274) there's a problem in highly scalable environments to avoid conflicts in storing dialogs in the same DB.
As @miconda pointed out:
> The internal id is made from two integers, hash entry and hash id (h_entry, h_id). h_entry is the index of the slot in the hash table used to store the dialog structure, computed by hashing over call id. h_id is an incremented integer specific for each hash table slot.
>
> So, when loading a new record from database, if its h_id is not conflicting with an existing dialog on the same hash table slot, all is ok, otherwise the module is not going to work properly with two dialogs having same h_id.
>
> So, when loading a new record from database, if its h_id is not conflicting with an existing dialog on the same hash table slot, all is ok, otherwise the module is not going to work properly with two dialogs having same h_id.
>
> Among solutions I thought of:
>
> 1. have the servers generating non conflicting h_id, by having a start value different per server and an increment step larger than the number of servers. Iirc, here was at some point a similar attempt, but somehow didn't make it. Let's say one has two servers, first server starts allocating h_id from 1 and increments by 2 (e.g: 1, 3, 5, ...) and the seconds start from 2 and increments with 2 (2, 4, 6, ...). Those values can be set via mod params, eventually with an option to rely on server_id.
> This should be the least intrusive in the other modules built on top of dialog. But it is rather rigid, with the example above, if one adds an extra server, it needs to reconfigure the old ones, so each server starts from either 1, 2 or 3 and increment by 3. Of course, one can set increment step to a larger value, like 100, and then has flexibility to deploy up to 1 hundred servers before having to reconfigure in case there is need for more server.
>
> 2. add server_id as the third field in the dialog id. It will require review of other modules using dialog and eventual code updates in those modules. One column to store the server_id needs to be added to dialog db tables.
>
> 3. switch from this conflicting id system to something like string unique ids, similar to what we have in usrloc records. This will require coding in other modules, changes to database schema, etc., so more work comparing with the above two, but could be the best in long term.
It would be great to have the option 3 developed if possible but we would also be satisfied with the option 2 to have a third field, server_id, in the dialog id.
Thank You!
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When K connects to another SIP proxy over TCP or TLS, the connection is shared for all requests to that destination. If the receiving SIP proxy is K, it means that only one worker process is handling all those requests. This may become a bottleneck if processing of requests is time consuming, e.g., because of DB operations.
It should therefore be possible one way or another to establish more than one parallel TCP or TLS connections to the same destination. One solution could be based on a function call, such as t_forward_connect(number), to be called before t_relay specifying the desired number of parallel connections shared randomly by outgoing requests.
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There are lots of module parameters that are not documented in the README file:
"amqp_timmer_process_interval"
"amqp_consumer_ack_timeout_micro"
"amqp_consumer_ack_timeout_sec"
"amqp_interprocess_timeout_micro"
"amqp_interprocess_timeout_sec"
"amqp_waitframe_timeout_micro"
"amqp_waitframe_timeout_sec"
"amqp_consumer_workers"
"amqp_query_timeout_micro"
"amqp_query_timeout_sec"
"pua_include_entity"
"json_escape_char"
"app_name"
"use_federated_exchange"
"federated_exchange"
"amqp_heartbeats"
"amqp_primary_zone"
"amqp_command_hashtable_size"
"amqp_result_avp"
"amqps_ca_cert"
"amqps_cert"
"amqps_key"
"amqps_verify_peer"
"amqps_verify_hostname"
Also, the follwing functions are not documented:
"kazoo_json_keys"
"kazoo_async_query"
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Hello,
I'm testing kamailio with a SIP client with presence support, when I get a connection drop from a client (i.e when the client lost the access to the Internet connection) I do not receive the change in the status to offline immediately, I just received when the timeout from the presentity database finishes.
I know that I'm able to change the expiry parameter from the presentity table, but that increases my battery consumption and the data exchange, the same happens if I reduce the publish interval on the client side.
The connection with the server are connected with the TLS or TCP.
@miconda Suggested that a solution could be:
"For TCP/TLS, what can work, it's to use tcpops module which can execute an event route when a tcp/tls connection is dropped. If you track the association of connection id with the call-id of the presence publish, then you may be able to do some tricks and lower the expires of the published presence state. I haven't had the time to see what's possible there, but could worth investing.
An alternative would be to enhance presence module to behave as registrar/usrloc (which can delete contact records on connection drop) to expire documents when tcp/tls connection is closed (not sure if anyone already added it)."
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From https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=843477
> From: OndÅej Surý <ondrej(a)debian.org>
> To: Debian Bug Tracking System <submit(a)bugs.debian.org>
> Subject: src:kamailio: stop using libval-dev for DNSSEC validation
>
> Package: src:kamailio
> Version: 4.4.3-2
> Severity: serious
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA512
>
> Dear maintainer,
>
> please stop using libdnsval-dev for DNSSEC validation. src:dnsval
> is very poorly upstream maintained (SVN repository, issue tracker,
> mailing list are non-functional) and I am going to schedule it for
> removal for stretch.
>
> In a long term you should migrated to better and maintained library
> (preferrably getdns or just libunbound), but now please just drop
> it.
>
> Cheers,
> Ondrej
>
> - -- System Information:
> Debian Release: 8.6
> APT prefers stable-updates
> APT policy: (500, 'stable-updates'), (500, 'proposed-updates'), (500, 'stable')
> Architecture: amd64 (x86_64)
>
> Kernel: Linux 4.2.0-41-generic (SMP w/24 CPU cores)
> Locale: LANG=en_DK.UTF-8, LC_CTYPE=en_DK.UTF-8 (charmap=UTF-8)
> Shell: /bin/sh linked to /bin/dash
> Init: systemd (via /run/systemd/system)
>
> -----BEGIN PGP SIGNATURE-----
So I'm going to remove **dnssec** module from the official Debian package
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Hi,
When using dialog module and track_cseq_updates when needing to update CSeq for authenticated invites, if early media is present the CSeq in the Rack header is not also incremented.
Kamailio version:
version: kamailio 4.4.2 (x86_64/linux) 892ad6
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_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: 892ad6
compiled on 11:18:36 Sep 28 2016 with gcc 5.4.0
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Sipwise hat ON:
We have more than one server and we need to keep things in db in order to share sca subscription/appearance between servers
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>From http://lists.kamailio.org/pipermail/sr-dev/2016-August/036231.html
On 24/08/16 10:17, Andreas Granig wrote:
> Hi,
>
> When translating strings with the dialplan module, you sometimes end up
> with very complex patterns for sanitizing the target string and
> performing some replacements, since for a given dialplan id, processing
> stops on the first match within the dialplan entries for a given dpid.
>
> The idea is to introduce a "continue" flag (default 0) in the dialplan
> table indicating to proceed with processing further dialplan entries
> within the given dpid after a match. The use case is to have a sanitize
> rule with highest priority which e.g. removes any spaces, dashes and
> parentheses from the target string, then with a lower priority do
> further processing with the cleaned up string, e.g. stripping leading
> double-zeroes or a plus when normalizing to E164 numbers, or replacing a
> single leading zero with the country code and so on.
>
> This is particularly useful when a user dials a number from an address
> book of some sort (e.g. on a mobile app) where number formats are stored
> with country-specific formating characters like "+1-(234)-567 890", and
> catching all of that in one rule results in really messy match/replace
> patterns.
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The local-request event route is not documented. When using our friend google I find an old blog post by @miconda but not much else. I can come up with some good text, but need to know what can be done in the route, like
- modify request uri
- add/remove headers
- drop request
- add/modify destination uri ($du)
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This will allow us to point https://www.kamailio.org/wiki/features/new-in-devel to the file and "force" developers to update that file with the new features when they are introduced in the repository
Remaining developers to upgrade the wiki seems pointless and anyone can create a PR for changes
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<!-- Kamailio Pull Request Template -->
<!--
IMPORTANT:
- for detailed contributing guidelines, read:
https://github.com/kamailio/kamailio/blob/master/.github/CONTRIBUTING.md
- pull requests must be done to master branch, unless they are backports
of fixes from master branch to a stable branch
- backports to stable branches must be done with 'git cherry-pick -x ...'
- code is contributed under BSD for core and main components (tm, sl, auth, tls)
- code is contributed GPLv2 or a compatible license for the other components
- GPL code is contributed with OpenSSL licensing exception
-->
#### Pre-Submission Checklist
<!-- Go over all points below, and after creating the PR, tick all the checkboxes that apply -->
<!-- All points should be verified, otherwise, read the CONTRIBUTING guidelines from above-->
<!-- If you're unsure about any of these, don't hesitate to ask on sr-dev mailing list -->
- [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
- [ ] Small bug fix (non-breaking change which fixes an issue)
- [x] New feature (non-breaking change which adds new functionality)
- [ ] Breaking change (fix or feature that would change existing functionality)
#### Checklist:
<!-- Go over all points below, and after creating the PR, tick the checkboxes that apply -->
- [x] PR should be backported to stable branches
- [x] Tested changes locally
- [ ] Related to issue #XXXX (replace XXXX with an open issue number)
#### Description
<!-- Describe your changes in detail -->
- It is allowed to use '@' within a username and/or a password for MySQL.
Database as a Service providers like e.g. Microsoft Azure use
these in usernames by default.
The existing URI parser always treated the '@' as a separator between
username:password and host:port/db parts of the URI.
The parser now checks how many '@' characters are present in the URI
and only treats the last occurence as separator for the host:port/db
part.
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/1346
-- Commit Summary --
* module_db_mysql: Allow '@' in username/password for db_url URIs
-- File Changes --
M src/modules/db_mysql/my_uri.c (35)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/1346.patchhttps://github.com/kamailio/kamailio/pull/1346.diff
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The module sipcapture provides the modparams "custom_field1_header" to "custom_field3_header", to store the content of up to three arbitrary header fields as separate columns in the database. They can be used as all other "standard" columns in homer.
However I found out that searching/filtering for even more (standardized, as well as private) header fields would be very helpful (I identified about ten interesting additional headers, but the number is constantly increasing ;-) ).
Could you thus please increase the number of custom fields?
(I suppose the most flexible solution could be something as the db_extra in the acc-module, which is not fixed to certain columns (regarding number of columns, as well as their names) (see https://www.kamailio.org/docs/modules/5.0.x/modules/acc.html#acc.p.db_extra ).)
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Below feature were discussed with Carsten Bock
Currently Diameter AVP values and especially Experimental Result Code is not available from the kamailio.cfg
Therefore I would like to request such kind of feature. I would like to use it mainly on Cx interface for Server-Assignment-Answer (SAA), Location-Info-Answer (LIA), Multimedia-Auth-Answer (MAA), Push-Profile-Answer (PPA)
It should be implemented such a way like in the ims_charging module.
Thank you very much in advance!
_Cheers,
Zoltan_
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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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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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(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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It seems a good idea to support JWT as a new SIP authorization method. Wonder if anyone is interested? Think auth_db would be the best spot to add support for JWT.
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#### Description
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192.0.0.0/29 is a reserved subnet for NATtin IPv4 across an IPv6 netwerk. For example Dual Stack Lite carrier grade NAT.
This is small change and updates the subnets used for NAT according to https://www.iana.org/assignments/iana-ipv4-special-registry/iana-ipv4-speciā¦
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/1488
-- Commit Summary --
* Support for
-- File Changes --
M src/modules/ipops/detailed_ip_type.c (1)
M src/modules/ipops/detailed_ip_type.h (2)
M src/modules/nat_traversal/nat_traversal.c (3)
M src/modules/nathelper/nathelper.c (1)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/1488.patchhttps://github.com/kamailio/kamailio/pull/1488.diff
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Tracking here the issue exposed by @linuxmaniac via PR #924, as it was merged and then reverted (by #1248), so anyone can be aware of as well as comment/propose solutions for it.
> Add "-fno-strict-aliasing" to compilation flags
>
> Fixes a slew of:
> "dereferencing type-punned pointer will break strict-aliasing rules"
> warnings from GCC for lines of the form:
> Py_INCREF(Py_True);
> and
> Py_INCREF(Py_False);
>
> due to the cast from PyIntObject* to PyObject*
>
> GCC is technically correct here; see:
> http://www.python.org/dev/peps/pep-3123/
> though this is unlikely to lead to non-working machine code.
The initial proposed patch for `src/modules/app_python/Makefile` is:
```
@@ -22,6 +22,8 @@ ifeq ($(OS), freebsd)
LIBS+=-pthread
endif
+# python2 https://www.python.org/dev/peps/pep-3123/
+DEFS+=-fno-strict-aliasing
DEFS+=-I${PYTHON_INCDIR}
DEFS+=-DKAMAILIO_MOD_INTERFACE
```
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