Dear all
one quick question, reading the module corex doc, seems that xflag are
message(transaction) flags. But I made a test and seems for some reason the
flag is not seeing activated at the onreply_route, when it's activated on
the request route. Seemed more like a script flag behaviour. Maybe I'm
missing something?
thanks a lot and regards
david
Hi Users,
I am new to kamailio, and trying to implement webrtc on kamailio, i am
getting following error when i try to login from tryjssip page.
Mar 29 21:11:15 kamailio kamailio[20570]: ERROR: <core>
[core/tcp_read.c:1535]: tcp_read_req(): bad request, state=7, error=4
buf:#012GET / HTTP/1.1#015#012Host: kamailio..com:7443#015#012Connection:
Upgrade#015#012Pragma: no-cache#015#012Cache-Control:
no-cache#015#012User-Agent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_15_2)
AppleWebKit/537.36 (KHTML, like Gecko) Chrome/80.0.3987.149
Safari/537.36#015#012Upgrade: websocket#015#012Origin:
https://tryit.jssip.net#015#012Sec-WebSocket-Version:
13#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language:
en-US,en;q=0.9,ur;q=0.8#015#012Sec-WebSocket-Key:
88fVCzdsvQPCrwV7gMrWaA==#015#012Sec-WebSocket-Extensions:
permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol:
sip#015#012#015#012#012parsed:#012GET / HTTP/1.1#015#012Host:
kamailio..com:7443#015#012Connection: Upgrade#015#012Pragma:
no-cache#015#012Cache-Control: no-cache#015#012User-Agent: Mozilla/5.0
(Macintosh; Intel Mac OS X 10_15_2) AppleWebKit/537.36 (KHTML, like Gecko)
Chrome/80.0.3987.149 Safari/537.36#015#012Upgrade: websocket#015#012Origin:
https://tryit.jssip.net#015#012Sec-WebSocket-Version:
13#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language:
en-US,en;q=0.9,ur;q=0.8#015#012Sec-WebSocket-Key:
88fVCzdsvQPCrwV7gMrWaA==#015#012Sec-WebSocket-Extensions:
permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol:
sip
I can see following error on chrome console.
WebSocket connection to 'wss://kamailio..com:7443/' failed: Connection
closed before receiving a handshake response
I tried to set tcp_accept_no_cl to value 'no' and this did not helped me
either.
Can someone please help me with the issue
Regards
Bilal Abbasi
Hi
I have a kamailio 5.1.2 as load balancer and registration offloading,
but I have a problem with the max tcp connections that it can handle.
I suspect that is a linux limit, but I don't find the reason or config.
When that limit arrives, I can't connect to kamailio and I receive
"Connection reset by peer", but I can't view any error message in the
logs.
If I check the connections in kamailio, I view that it have "free" connections:
# kamctl kamcmd core.tcp_info
{
readers: 8
max_connections: 4096
max_tls_connections: 2048
opened_connections: 2655
opened_tls_connections: 0
write_queued_bytes: 0
}
I have this configs in kamailio.conf (related to tcp)
disable_tcp=no
tcp_connection_lifetime=3610
tcp_connect_timeout=5
tcp_crlf_ping=yes
tcp_accept_aliases=no
tcp_keepalive=yes
tcp_keepidle=5
tcp_rd_buf_size=65536
tcp_conn_wq_max=131072
mlock_pages=yes
shm_force_alloc=yes
tcp_max_connections=4096
The shm memory to 256 and the pkg memory to 32.
And, following this doc:
https://github.com/kamailio/kamailio/blob/master/doc/tutorials/tcp_tunning.…
I have setted this values:
net.ipv4.ip_local_port_range = 1024 65535
net.core.somaxconn = 65535
net.core.netdev_max_backlog = 182757
Also, I had checked the limits for the main process pid:
Limit Soft Limit Hard Limit Units
Max cpu time unlimited unlimited seconds
Max file size unlimited unlimited bytes
Max data size unlimited unlimited bytes
Max stack size 8388608 unlimited bytes
Max core file size unlimited unlimited bytes
Max resident set unlimited unlimited bytes
Max processes unlimited unlimited processes
Max open files 1048576 1048576 files
Max locked memory 16777216 16777216 bytes
Max address space unlimited unlimited bytes
Max file locks unlimited unlimited locks
Max pending signals 386297 386297 signals
Max msgqueue size 819200 819200 bytes
Max nice priority 0 0
Max realtime priority 0 0
Max realtime timeout unlimited unlimited us
The service is running inside a lxc container, without any resource
limit, connected to the outside word throught macvlan interface.
Where can I find problem source?
Best regards
Hello,
I'm using RTPproxy for the first time in bridged mode and I can't get kamailio/rtpproxy to rewrite the c parameter to the correct public ip address of kamailio.
The setup is as following:
Carrier ------[fiber]------ Kamailio ---------[lan]--------- Freeswitch
Kamailio is listening on two interfaces:
1) Private: 172.0.0.1
2) Public: 192.168.0.1 (since we have a dedicated fiber with our carrier, this is its public address)
Freeswitch is listening on:
1) 172.0.0.2
Carrier is on:
1) 10.0.0.1
I've started an rtpproxy instance on the Kamailio box using:
rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.1 172.0.0.1
I've played around with rtpproxy_manage() and the various flags (ie, ei), but I can't get kamailio to set the correct public IP when the 200 OK has to be sent back to the carrier.
It always sets it to its private address, instead of its public address.
I'm using Kamailio 4.2 with sippy/rtpproxy 2.0.
Could someone please point me into the right direction?
Thanks!
Grant
Hello,
wondering if anyone here is aware of a lightweight sip app that can
answer a call, play some file and/or do echo mode, mainly targeted at
using it for basic sip routing and call testing. Of course I know that
Asterisk and FreeSwitch (or even SEMS) can do that, but they have many
dependencies, requiring quite some resources to run them, so I thought
maybe someone here figured out different solutions, eventually cli based
apps like pjsua or baresip. GUI apps for Linux are also fine if they can
be configured for such behaviour.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Hi All,
Just curious to ask as how to save $dlg_var for a multi-leg/branched call.
When I call a subscriber with multiple Contacts, and one branch answer
while rest are CANCEL'd from that point onward I've seen that the dialog
and its variables disappear from Kamailio memory and DB.Is this how its
intended to be ?
Here are the relevant debug line:
DEBUG: dialog [dlg_hash.c:1266]: next_state_dlg(): dialog 0x7f7c1461e2e8
changed from state 2 to state 5, due event 4 (ref 2)
DEBUG: dialog [dlg_handlers.c:574]: dlg_onreply(): dialog 0x7f7c1461e2e8
failed (negative reply)
DEBUG: dialog [dlg_cb.c:271]: run_dlg_callbacks(): dialog=0x7f7c1461e2e8,
type=4
DEBUG: dialog [dlg_handlers.c:1050]: dlg_set_tm_waitack(): registering TMCB
to wait for negative ACK
Best Regard,
Sammy
Hello,
I'm using Kamailio MSRP module as a MSRP relay, It works successfully when
I send one-to-one MSRP messages. I am having trouble while sending msrp
messages to multiple receivers. My question is that can I use this module
to forward msrp messages not just one receiver, but multiple receivers?
Can this module be used like MSRP switch?
Thanks in advance.
Description
I'm running Kamailio 5.2.0, whenever I relay an invite via Kamailio, my original contact header is changed from the original:
sip:+XXXXXXXXX@YYY.YYY.YYY.YYYY:5060;transport=udp;gw=netvision
To
sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*.
Even when I set it correctly on my route, using:
remove_hf("Contact");
append_hf("Contact: sip:$tU@YYY.YYY.YYY.YYY:5060;transport=udp;gw=netvision\r\n", "Contact");
It still ends up being modified.
What can I do to keep the contact header as it is?
SIP Traffic
U YYY.YYY.YYY.YYY:5060 -> 81.24.193.248:5060
INVITE sip:+442033202609@81.24.193.248:5060 SIP/2.0.
Record-Route: <sip:YYY.YYY.YYY.YYY;r2=on;lr=on;ftag=mp0S9yH11vryH;vsf=AAAAAAAAAAAAAAAAAAAAAAADCAEACRgPGB8AAy4xNjM->.
Record-Route: <sip:2YYY.YYY.YYY.YYY;line=sr-.n274V8TlidQMVyUlg87vV8nlfDXlVHB9VD0.VHB9VDoLGtNRbH7ltlTEkyQl3jXEky1LNMoRkt5ukt5ukt5ukt5ukt5ukt5ukt5ukt5W8M5WktckoLuWAsduktTMm64pCAD>.
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK53e9.308596b23ed683368534c9d609dce0f6.0.
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKsr-kA2uvfsUlerJWtIylgCXvV8n4edQlgIUlgKQ4Vk74gI1.oJVjb2njburlgCXvV8n4edQlgIUlgKQ4HWsRbWPLVDX.GrXLcAHlcC74nZXKbTVpcHh4b6mMGZvJgLsLcJTkgtGu2PGgy**.
Max-Forwards: 67.
From: +18702935016 <sip:+18702935016@YYY.YYY.YYY.YYY>;tag=mp0S9yH11vryH.
To: <sip:+442033202609@81.24.193.248>.
Call-ID: !!:jbkfMgyHjbsDjblfjqAQlVldvgyHKolDlcIHlcknKVIdjcIT.
CSeq: 18182498 INVITE.
Contact: <sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*>.
User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20180706T160334Z~de3df8dc0e~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 224.
Remote-Party-ID: "+18702935016" <sip:+18702935016@YYY.YYY.YYY.YYY>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1585439298 1585439299 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ.
s=FreeSWITCH.
c=IN IP4 ZZZ.ZZZ