Hey all,
I've been using the PHONENUM module, which uses Google's libphonenumber
library, to perform number validation among other handy things within the
route script. One problem I've run into however is the metadata in the
latest stable (5.4) repo is pretty stale.
I'm curious if anyone is running this module, and if so how they've gone
about updating the metadata used by the underlying library. If I have to
ditch the repo and compile everything manually, that's not a huge deal.
Just hoping there's another way.
Thanks!
Zach
Hi all
Just installed a kamailio 5.4.1 for a simple proxy scenario.
I created the version,address and trusted tables, and added some ips to the
trusted via kamctl.
If I execute "kamctl trusted show" I see the list of servers. But with "kamctl
trusted dump" I receive
{
"jsonrpc": "2.0",
"error": {
"code": 500,
"message": "No trusted table"
},
"id": 13063
}
when I start kamailio with debug I see these messages:
b_do_submit_query(): submitted query: select `table_version` from `version`
where `table_name`='trusted', result 0
my version table looks like this:
+----+------------+---------------+
| id | table_name | table_version |
+----+------------+---------------+
| 1 | version | 1 |
| 2 | trusted | 6 |
| 3 | address | 6 |
+----+------------+---------------+
any hints?
cheers,
Jon
--
PekePBX, the multitenant PBX solution
https://pekepbx.com
White testing WebRTC, I noticed that TLS connection from Chrome browser
85.0.4183.121 to Kamailio's (master version) TLS listening port fails.
Chrome sends Client Hello and K responds with Server Hello. After that
Chrome terminates the handshake by sending TCP FIN.
Connection setup from Firefox works fine. After receiving Server Hello,
is sends Change Cipher Spec, etc.
My K is using Let's Encrypt server certificate and is server tls_method is
TLSv1.2+.
Any ideas what goes wrong? Has anyone else experienced this?
-- Juha
Dear Users
How can I increase private memory in the Kamailio container?
the service was started with the following command
docker run -d --restart unless-stopped --network host -v /root/volumes/scscf/:/etc/kamailio --name scscf -e IP=10.161.224.90 -e INT_IP=10.161.224.90 10.161.224.196:5000/kamailio-ngvoice
PS: I don't have a docker file to build the image again.
Regards
Hamid R. Hashmi
Hello there,
Suddenly the usrloc replication through DMQ stopped, but the interesting
thing is that DMQ cluster status is OK, htable replication still working
only usrloc_dmq stopped working.
I increased the kamailio loglevel to 3 in order to see if there was some
log related and then registered a peer, but the only DMQ logs that I see
are related with notification_peer and htable replication, nothing related
with usrloc_dmq.
What I can do more here to try to understand what went wrong?
I haven't restarted the cluster yet, because I would like to try to
understand the root cause of this issue.
Any advice would be appreciated
Thank you
--
Cumprimentos
José Seabra
Hi all,
Today I saw the following errors reported by rtpengine module:
Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) DEBUG: rtpengine
[rtpengine.c:2453]: rtpp_function_call(): proxy reply:
d6:result5:error12:error-reason15:Unknown call-ide
Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) ERROR: rtpengine
[rtpengine.c:2474]: rtpp_function_call(): proxy replied with error: Unknown
call-id
This was on the 200 OK response back from our Freeswitch.
Any idea what would cause this?
Thanks.
--
Andy Chen
Sr. Telephony Lead Engineer
achen@ <achen(a)thinkingphones.com>fuze.com
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Hi,
We are using Jitsi Meet for Video Conference Service. We want our SIP
endpoint to join in the conference room in Jitsi. For that i am trying to
configure Kamailio SIP Server in our environment. I have installed the
Server and my endpoints are registered in my SIP server. Both endpoints are
able to connect to each other. But I am unable to configure the SIP
endpoints to join in the Meet Conference Room. Please guide me to configure.
Thanks and regards,
Santanu
Hello,
Kamailio *sdpops *module consist of several useful functions for
manipulating SDP on Proxy side such as *sdp_keep_codecs_by_name*() .
This function *sdp_keep_codecs_by_name*("PCMU,PCMA,GSM") finds codecs id
[0,8,3] and if incoming packet consist of other than these codecs, they are
removed ! For example, assume incoming packet SDP --> [8 0 3 9 101] after
applying this function 9 and 101 are removed and outgoing packet SDP
consist only PCMU,PCMA,GSM [0,8,3].
This function does taking given codec *precedence/priority* into account.
As you see above message SDP is 8 0 3, different priority than inputted 0 8
3. I need to change *order *of codecs too ! Would you help me to do it ? I
couldn't find *add *operation on *sdpops *module
Regards,
Melek
Wondering if this is intended behavior. We have a customer doing some
craziness with the TO header. They are including their tech prefix in the
TO header as well as the RURI. It doesn't cause any issue in our systems
but apparently it does with one of our vendors. In an effect to prevent
the issue, we thought we would check if the tech prefix was added on the TO
header and remove it.
Initially, I tried this by changing the $tU variable.
$avp(tp_len) = $(avp(techprefix){s.len}) + 1; # Get the Length of the Tech
Prefix and add 1 for the # or *
$tU = $(tU{s.strip, $avp(tp_len)});
The result ended up with the duplication of the DNIS: sut <sip:911999#
1812555111118125551111@172.16.3.45:5060>
Where 911999# was the tech prefix and 18125551111 is the DNIS which was
still an issue for the vendor.
Through additional reading, I found the suggestion to use
the uac_replace_to function in the UAC module which I implemented as
follows:
$avp(new_to_hdr_uri) = $(tu{re.subst,/sip:(.*[\*#])(.*)/sip:\2/});
uac_replace_to("$avp(new_to_hdr_uri)");
This resulted in the original TO header username to be appended at the end
of the uri: sut <sip:18125551111@172.16.3.45:5060911999#18125551111>
Wondering if I am doing something wrong or this is the way these functions
are designed to protect the TO header contents for Dialog matching.
--
Sam D Ware
