Hello All,
I have had an issue for quite some time where I needed to open two UDP ports (via LISTEN) for some softphone UAC's to stay connected. Funny enough there is one Provider that works with the Single UDP. I think I finally see why I was needing two listening UDP ports. When the ACK is returning from Asterisk to Kamailio, the ACK is coming back as protocol TCP, not SIP, causing the call to drop. I finally realized this but I am not sure what would cause this and why one provider phone works but not the others. Once this is resolved, I may actually say...It works!
Thank you,
-Steve
For some reason, Voicemail Notify messages coming from Asterisk Boxes - are not forwarded to the SIP Extensions by Kamailio.
This is the SIP Packet coming to the Kamailio:
2018/08/28 10:59:19.165644 1.1.1.1:5060 -> 2.2.2.2:5060
NOTIFY sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK5f628579;rport
Max-Forwards: 70
Route: <sip:2.2.2.2:5060;transport=tcp;ftag=0472e823;lr=on;nat=yes>
From: "asterisk" <sip:asterisk@1.1.1.1>;tag=as1833c329
To: <sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2>;tag=0472e823
Contact: <sip:asterisk@1.1.1.1:5060>
Call-ID: 93183ZjE3YmI0OGYxMDNiMzhlNzVjZjRjNTc2NTMwZGNmYzk
CSeq: 103 NOTIFY
User-Agent: NUACOM
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93
Messages-Waiting: no
Message-Account: sip:asterisk@1.1.1.1
Voice-Message: 0/0 (0/0)
And this is the message which I'm receiving in the Kamailio logs:
[RELAY] Error when forwarding Request
ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL)
This is the Relay route:
# Forwarding Statefully
if ( ! t_relay()) {
xlogl( "L_ERR" , "[RELAY] Error when forwarding Request \n " );
sl_reply_error();
}
exit ;
Hello,
I am using Kamailio as a SIP proxy.So it receives SIP packet from internet
and forwards it to FS servers in local network.When i execute "ss" command
i see very high value in RECV-Q column.I THINK IT IS NOT NORMAL.PLEASE
CORRECT ME IF I AM WRONG.
====================================================
[root@fep-1 proc]# ss -u -a -n -e | grep 5060
UNCONN 0 0 10.50.8.1:5060 *:*
ino:6831630 sk:fd <->
UNCONN 0 0 10.50.7.18:5060 *:*
ino:6831629 sk:fe <->
UNCONN *1183104* 0 10.50.7.254:5060 *:*
ino:6831627 sk:ff <->
UNCONN *84864* 0 2607:f900:1:3::254:5060
:::* ino:6831628 sk:100 v6only:1 <->
======================================================
Initially i thought that there is something in script which must be causing
kamailio to process UDP request slower but i faced same issue with a very
simple script where i simply reply with stateless 200 OK for each sip
request :
request_route {
sl_send_reply("200","OK");exit;
$avp(uuid) = $rm + "-" + $ci;
........
.........
.........
.......
Server configuration :
OS : CENTOS 7
Kernel : 4.16
CPU : 5 X Intel(R) Xeon(R) CPU E5-2680 v2 @ 2.80GHz
RAM : 32 GB
Please help me debugging this issue.Thanks in advance
Hi All,
I am trying get the 10digit number called in on using the following
$var(caller) = $(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]});
I thought I could use a regular expression for the expression in re.subst,expression
Thank you,
-Steve
Hi Joel,
Actually, I have a strange issue. If I have a single UDP IP:Port listening, most Providers/Phones have two-way Audio/Video, however, with some Providers/Phones, I need to have two UDP IP:Port listners in order to get two-way Audio/Video. Here is the strange thing. If I enable two, the Providers/Phones that worked when I had a single listener only have one-way Audi/Video.
I use Kamailio 5.1 and Asterisk 15.3 (pjsip). This behavior is so strange,
Thank you,
-Steve
Hi Florian,
Thank you for your response. I checked and direct-media is off. Just as a recap, here is where I am
I just can't get rtpengine to work. I have tried multiple configurations, but to no avail. Note that calls work good if rtpengine is disabled.
Here is my setup =>
Public IP: 20.20.20.20
Private IP 10.10.10.10
flow =>
webrtc client <-> kamailio+rtpengine <-> asterisk <-> kamailio <-> legacy sip phone
rtpenngine startup (I have tried a few different startups) =>
rtpengine --interface=int/10.10.10.10 --interface=ext/10.10.10.10\!20.20.20.20 --listen-ng=127.0.0.1:12221 --pidfile=/var/run/rtpengine --dtls-passive -f -m 10000 -M 20000 -E
kamailio =>
Invites: rtpengine_manage("trust-address replace-origin replace-session-connection direction=ext direction=int ICE=remove RTP/AVP");
Reply's: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF");
I have tried direction ext ext; and many other combinations, each producing its own incorrect behavior.
Thanks again,
Steve
