(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
>From what I experience dynamic routing (drouting) module is not supporting SRV resolution if NAPTR is turned off (dns_try_naptr=no). Is this intentional to not have SRV if NAPTR is turned off? Or some special format of entry is expected for SRV records in gateways table? I've tried with just **domain** and **_sip._udp.domain** but it it the same.
#### Reproduction
1. Use Dynamic routing module
2. set dns_try_naptr=no
3. insert gateway with SRV domain name in dynamic routing gateways table.
4. execute kamcmd drouting.reload
5. in the log of the kamailio is the error: ERROR: drouting [routing.c:337]: add_dst(): cannot resolve <domainexample>
### Possible Solutions
Upon reload of drouting tables, dynamic routing tries to add destinations and resolve them:
https://github.com/kamailio/kamailio/blob/08a9ed3d136c861f990e4f63c2c0544e0…
But, since port that is sent is NULL, SRV is being skipped later on. It seems that SRV would be executed if port would be 0 instead of NULL.
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
version: kamailio 5.1.9 (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 4.9.2
```
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Hi, Can someone help me?
I self host a kamailio using my raspberry pi as a load balancer for my two
asterisk servers and get a did number. when I call to my DID number it
points to my kamailio and kamailio will distribute to asterisk server but
the call has no audio. I tried port forwarding ports 5060 for SIP and
10000-20000 for RTP but it still does not work.
Any help is much appreciated. Thank you in advance