We have a SIP doorstation Grandstream GDS3710. When I connect it via
kamailio, it does not setup call correct.
Kamailio sends a 200 OK with this to field:
t:
<sip:0001@192.168.1.150:5069>;tag=e33a13d3-c18c-4df1-b68f-446e42957de6
Grandstream answers with:
To: <sip:0001@192.168.1.150:5069>
As the code follows this example:
https://kamailio.org/docs/modules/4.3.x/modules/dispatcher.html#dispatcher.…,
it seems obvious that this will fail, as there is not
route[WITHINDLG] {
if (has_totag()) {
There simply is not to tag. Is this a bug in the grandstream?
If so how could that go unnoticed, or is it me who misses something?
I have attached the full sip text for the 200OK and the ACK
--
-------------------- Med Liberalistiske Hilsner ----------------------
Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk
Hello group,
The call flow is:
useragent (TLS) -> Kamailio -> Sems Server.
During the establishment of a call (before the 200OK or during a
retransmission of the 200OK issued by Sems Server via Kamailio),
if a user agent (TCP) loses its connection (the user agent is on a mobile
network for example or the network is not reliable.) and sends a "register"
to Kamailio (same source IP address but a different source TCP port), is it
possible for Kamailio to take into account this new TCP port with the
function t_relay () ?
Abdoul
Hello,
We had memory leak on a production server and to find it we removed optional modules one by one.
As result we found that problem is in the sanity module.
Unfortunately I don't have time to fix it since I am going to vacation.
We used the following checks: ruri_sip_version (1), required_headers (4), cseq_method (32), cseq_value (64), cotent_length (128), expires_value (256), proxy_require (512), parse uri's (1024).
So be aware that this module is not ready to use in production.
Best regards,
Konstantin
hi configured websockets on kamilio 5.1.3
I am trying to register the client based on webrtc client ,(sip.js)
the registration is fine but i am not getting the correct address if i
run "*kamctl ul show"*
* In the address filed I want the sip:<id>@<ip>:<port>ex:
sip:12345@127.0.0.1:12345 <http://sip:12345@127.0.0.1:12345> *
output shows:
{
"jsonrpc": "2.0",
"result": {
"Domains": [{
"Domain": {
"Domain": "location",
"Size": 1024,
"AoRs": [{
"Info": {
"AoR": "1xxxxxxxxx",
"HashID": 1xxxxx,
"Contacts": [{
"Contact": {
*"Address":
"sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws"*,
"Expires": 397,
"Q": 1,
"Call-ID": "ma4feu2i43bnu6ophe4k78",
"CSeq": 86,
"User-Agent": "SIP.js/0.5.0-devel BB",
"Received": "sip:X.X.X.X:56087;transport=ws",
"Path": "[not set]",
"State": "CS_NEW",
"Flags": 0,
"CFlags": 192,
"Socket": "tcp:X.X.X.X:8080",
"Methods": 783,
"Ruid": "uloc-5b116457-1dc1-03",
"Instance":
"<urn:uuid:0e879ebf-0a31-45a8-9fd9-6b937d3328d4>",
"Reg-Id": 1,
"Server-Id": 0,
"Tcpconn-Id": 259,
"Keepalive": 1,
"Last-Keepalive": 1527876355,
"Last-Modified": 1527876355
}
}]
}
}
],
"Stats": {
"Records": 1,
"Max-Slots": 1
}
}
}]
},
"id": 18958
}
In the address filed I want the sip:<id>@<ip>:<port>
ex: sip:12345@127.0.0.1:12345
Please give me any suggestions on this, which webrtc client can i use to
register to kamailio and able to receive on the client and to make outbound
calls from the client.
--
Thanks and regards
Vinod.M.N
Hey all,
I'm looking to put Kamailio behind a TCP load balancer that is SIP-unaware.
My application is deployed in AWS and I'm tying to place Kamailio behind an
ELB.
For the most part, everything is fine. For my specific implementation I'm
disabling UDP as a signaling transport and using only TLS. This enables me
to not have to worry about a SIP-aware LB at the edge because replies to an
incoming request will be sent over the existing established TCP socket
(avoiding any crazy routing requirements).
However - this poses an issue with source addresses. Does Kamailio support
anything like the proxy protocol (
http://www.haproxy.org/download/1.5/doc/proxy-protocol.txt) for getting TCP
stream information from a load balancer? Do I need to go back to exposing
it directly to the world so that I can get source addresses?
Thanks,
Colin
Hello Vinod,
We used Doubango sipML5 web phone
https://www.doubango.org/sipml5/call.htm?svn=252
It was quite good for debugging and testing webrtc calls.
Most of work in kamailio config I made by using that client.
Best regards.
Konstantin
>
>Message: 8
>Date: Fri, 1 Jun 2018 23:50:50 +0530
>From: vinod mn < vinodmn.dbit(a)gmail.com >
>To: sr-users(a)lists.kamailio.org
>Subject: [SR-Users] configuring websockts in kamilio
>Message-ID:
>< CAHeWXy7h4Tcp4Sg9_eM3teCHBdJ5bwHa9TYSJy+7P4e+bm1hqQ(a)mail.gmail.com >
>Content-Type: text/plain; charset="utf-8"
>
>hi configured websockets on kamilio 5.1.3
>I am trying to register the client based on webrtc client ,(sip.js)
> the registration is fine but i am not getting the correct address if i
>run "*kamctl ul show"*
>
>
>* In the address filed I want the sip:<id>@<ip>:<port>ex:
>sip:12345@127.0.0.1:12345 < http://sip:12345@127.0.0.1:12345 > *
>output shows:
>{
> "jsonrpc": "2.0",
> "result": {
> "Domains": [{
> "Domain": {
> "Domain": "location",
> "Size": 1024,
> "AoRs": [{
> "Info": {
> "AoR": "1xxxxxxxxx",
> "HashID": 1xxxxx,
> "Contacts": [{
> "Contact": {
> *"Address":
>"sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws"*,
> "Expires": 397,
> "Q": 1,
> "Call-ID": "ma4feu2i43bnu6ophe4k78",
> "CSeq": 86,
> "User-Agent": "SIP.js/0.5.0-devel BB",
> "Received": "sip:X.X.X.X:56087;transport=ws",
> "Path": "[not set]",
> "State": "CS_NEW",
> "Flags": 0,
> "CFlags": 192,
> "Socket": "tcp:X.X.X.X:8080",
> "Methods": 783,
> "Ruid": "uloc-5b116457-1dc1-03",
> "Instance":
>"<urn:uuid:0e879ebf-0a31-45a8-9fd9-6b937d3328d4>",
> "Reg-Id": 1,
> "Server-Id": 0,
> "Tcpconn-Id": 259,
> "Keepalive": 1,
> "Last-Keepalive": 1527876355,
> "Last-Modified": 1527876355
> }
> }]
> }
> }
> ],
> "Stats": {
> "Records": 1,
> "Max-Slots": 1
> }
> }
> }]
> },
> "id": 18958
>}
>
>In the address filed I want the sip:<id>@<ip>:<port>
>ex: sip:12345@127.0.0.1:12345
>
>Please give me any suggestions on this, which webrtc client can i use to
>register to kamailio and able to receive on the client and to make outbound
>calls from the client.
>
>--
>Thanks and regards
>Vinod.M.N
>
Hello guys,
I setup the CDRs according to the instructions on
https://kb.asipto.com/siremis:install40x:accounting
I get CDRs nicely, but the cdrs, even though they have a src_ip, there's no
dest_ip, I mean I need the IP where the call was forwarded to...
What do i need to do to get that in the cdr?
Thanks!
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
ᐧ
Hello List users:
I am trying to upgrade to the current version of Kamailio from the older
4.4 version to the current version. The 4.4 is functional but there are
some features of 5.1 that we wanted to test with.
It seems that my version of Linux (RHEL 6.9 or Oracle 6.9) has an issue
with a library file.
Any help would be useful.
Thank you.
Kevin
Notes:
I used the "yum update kamailio" function.
(edited for clarity - it showed a long list of modules to be updated)
"--> Finished Dependency Resolution
Error: Package: kamailio-http-utils-4.4.7-1.1.x86_64
(@home_kamailio_v4.4.x-rpms)
Requires: kamailio = 4.4.7
Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms)
kamailio = 4.4.7-1.1
Updated By: kamailio-5.1.3-1.el6.x86_64
(home_kamailio_v5.1.x-rpms)
kamailio = 5.1.3-1.el6
Error: Package: kamailio-http-utils-4.4.7-1.1.x86_64
(@home_kamailio_v4.4.x-rpms)
Requires: libkmi.so.1()(64bit)
Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms)
libkmi.so.1()(64bit)
Updated By: kamailio-5.1.3-1.el6.x86_64
(home_kamailio_v5.1.x-rpms)
Not found
Error: Package: kamailio-xml-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms)
Requires: kamailio = 4.4.7
Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms)
kamailio = 4.4.7-1.1
Updated By: kamailio-5.1.3-1.el6.x86_64
(home_kamailio_v5.1.x-rpms)
kamailio = 5.1.3-1.el6
You could try using --skip-broken to work around the problem
You could try running: rpm -Va --nofiles --nodigest"
However the "missing" library is there in /usr/lib64.
[root@sipserver tmp]# find / -name libkmi.so.1
/usr/lib64/kamailio/libkmi.so.1
/usr/lib64/libkmi.so.1
/lib64/libkmi.so.1
To provide additional help, See:
[root@sipserver tmp]# kamailio -version
version: kamailio 4.4.7 (x86_64/linux) 35d8ab
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: 35d8ab
compiled on 15:38:14 Feb 26 2018 with gcc 4.4.7
Hi,
I configured the websocket in kamailio and it is running properly, which is
the best webrtc client (with proper code as I dont't want to spend more
time in writing the codes based on the example snippets ) can I use for
registering, and making/ receiving the calls on the webrtc-client .
requirement:
webrtc-client(??)->register-> kamailio(already configured to use websocket)
webrtc-client -> makecall to kamailio
webrtc-client receives call from another client(sip not webrtc-client)
--
Thanks and regards
Vinod.M.N
