Hi,
I'm using Kamailio with presence enabled and Asterisk PJSIP and
outbound-publish. My problem is happening when I place 2 consecutive calls
from Asterisk :
When I make a first call Asterisk sent the following:
PUBLISH sip:201@192.168.100.37 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.37:5080;rport;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e69723a610
82
From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf
To: sip:201@mydomain.com
Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183
CSeq: 10697 PUBLISH
Event: dialog
Expires: 180
Max-Forwards: 70
User-Agent: Asterisk PBX 14.6.0
Content-Type: application/dialog-info+xml
Content-Length: 247
<?xml version="1.0" encoding="UTF-8"?> early..
Kamailio replies :
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.100.37:5080;rport=5080;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e6972
3a61082;received=192.168.100.37
From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf
To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-ff39
Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183
CSeq: 10697 PUBLISH
Expires: 180
SIP-ETag: a.1518775074.19863.16.0
Server: kamailio (5.0.5 (x86_64/linux))
Content-Length: 0
When the call is done, Asterisk sent another PUBLISH telling that the call
if terminated :
PUBLISH sip:201@192.168.100.37 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.37:5080;rport;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97ab8c7
52
From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce
To: sip:201@mydomain.com
Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183
CSeq: 10698 PUBLISH
Event: dialog
SIP-If-Match: a.1518775074.19863.16.0
Expires: 180
Max-Forwards: 70
User-Agent: Asterisk PBX 14.6.0
Content-Type: application/dialog-info+xml
Content-Length: 230
<?xml version="1.0" encoding="UTF-8"?> terminated..
And Kamailio replies :
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.100.37:5080;rport=5080;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97
ab8c752;received=192.168.100.37
From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce
To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-48b4
Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183
CSeq: 10698 PUBLISH
Expires: 180
SIP-ETag: a.1518775074.19873.18.1
Server: kamailio (5.0.5 (x86_64/linux))
Content-Length: 0
Here, the SIP ETag is a.1518775074.19873.18.1.
The problem is if I make a new call before the expiration of the previous
SUBSCRIBE, Asterisk reuse this SIP ETag according to the RFC :
PUBLISH sip:201@192.168.100.37 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.37:5080;rport;branch=z9hG4bKPj9d13bb82-31d9-48db-9672-bd4b6b4f22
f0
From: sip:201@mydomain.com;tag=33e6b028-0444-4b3a-8bc2-4a987a291528
To: sip:201@mydomain.com
Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183
CSeq: 10699 PUBLISH
Event: dialog
SIP-If-Match: a.1518775074.19873.18.1
Expires: 180
Max-Forwards: 70
User-Agent: Asterisk PBX 14.6.0
Content-Type: application/dialog-info+xml
Content-Length: 247
<?xml version="1.0" encoding="UTF-8"?> early.
Kamailio refuse it with this error : "Trying to update an already terminated
state. Skipping update." because the call is considered as terminated.
The RFC is stating :
When updating previously published event state, PUBLISH requests MUST
contain a single SIP-If-Match header field identifying the specific
event state that the request is refreshing, modifying or removing.
This header field MUST contain a single entity-tag that was returned
by the ESC in the SIP-ETag header field of the response to a previous
publication.
Why Kamailio is acting like that?
Best regards,
Cyrille
Hi!
I'm working with kamailio and rtpproxy.
When I configure the clients to send sip messages through TCP adding the
";transport=tcp" in the sip message the rtpproxy is not working anymore.
The code is never passing the point pasted below, inside route[NATMANAGE]
in kamailio.cfg
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;
Therefore, rtpproxy_manage("co"); is never reached.
Could it be a parsing problem? ( check_route_param("nat=yes") )
The config file is exactly the same as the one provided in the kamailio
repository (the default one)
Thanks!
Hi people!
After a bit of research and lib installations, I've managed to build the
full kamailio deb packages for a raspberry pi 3 (version B) running
raspbian stretch (lite) that can be found at [1] (with a really few module
exceptions that might be used in really specific scenarios -> I think)
Just for fun, I'm planning to do some CPS stats tomorrow (if I got the
time), with a basic config and also share here, but, if someone else tried
it before and is able to share some stats/benchmarks, would be greatly
appreciated (and interesting to compare, given my "decent" rpi overclock),
because I'm also curious about VoIP stuff on raspberry pi! :D
[1] https://github.com/smititelu/rpi/tree/master/kamailio/5.1.1
Thank you,
Stefan
Hi everyone,
we're valuating on rewriting our kamailio routes in python.
We're doing a lot of string manipulation and GraphQL API queries so
the kamailio scripting language is a little bit limited for this use
case.
Can someone tell us if the KEMI framework is stable enough for production?
Kind regards,
--
Aleksandar Sosic
mail: alex.sosic(a)timenet.it
skype: alex.sosic
cell: +385 91 2505 146
Hello,
I'm trying to use this configuration:
Softphone A audio codec g722 and Opus
Softphone B audio codec PCMA and PCMU
Kamailio with rtpengine_manage with codec-transcode-PCMU
codec-transcode-PCMA in the request
Softphone A call Softphone B
The RTPEngine DEBUG show many Decoder Error:
https://pastebin.com/k1Zhw8HS
Thank you
Regards
---
I'm SoCIaL, MayBe
El 13/02/2018 a las 20:05, Richard Fuchs escribió:
> On 2018-02-13 07:22 PM, Social Boh wrote:
>> Thank you.
>>
>> which is the best way to implement ir
>>
>> codec-strip-all and the add codecs?
>>
>> Is there a document about how work the transcoding?
> It's really up to you and depends on what you want to achieve. Doing a
> strip-all and then adding codecs you want to offer is definitely one
> way. Another is to just add codecs that you know or think the peer may
> want to see.
>
> Cheers
>
5.1.1: if python_exec method returns 0, then the script hangs
If the python_exec method return -1, then expressions treat it as if
it returned zero.
Somewhere there seems to be an offset by 1 issue.
if (!python_exec("method")) {
# if you return -1, the script does not hang but treated like zero
}
##################
Example 1:
class kamailio(object):
def hello(self, msg):
KSR.info("hello, world\n")
return 0
Calling: python_exec("hello")
will cause script hang
###################
Example 2:
class kamailio(object):
def hello(self, msg):
KSR.info("hello, world\n")
return -1
Calling python_exec("hello")
if (!python_exec("hello")) {
## method returns -1
## why did we get here?
xlog("L_INFO","method return zero!\n");
}
shows INFO: <script>: method return zero!
Using 5.1, the python_exc args string is getting an extra [_0' appended
When I call python_exec("method", "string");
In python:
def method(self, msg, args):
KSR.info("DBG: {} {} {}".format(msg, args))
This will show 'string 0'; instead of string.
AAlba
Hi List,
I just read on the RTPEngine page the Proxy RTP now support transcoding:
https://github.com/sipwise/rtpengine#transcoding
is there a plan to support it?
Regards
--
---
I'm SoCIaL, MayBe
