With alias behind me I want to tackle a simple sequential and simulring (parallel). I see docs on using lookup() for simul and t_load_contacts but I don't see the basics on what table needs to be populated to get this working.
Test set is basic :
1. 1234567890 from carrier2. Hunt to 201(a)mytestdomain.com, 202(a)mytestdomain.com. These are both rgistered.
What tables do I populate or what CLI tools do I use to get this working.
Thanks
KD
Hi Team,
I have setup IM server with kamailio using MSILO module.
Its working fine now.
I have a scenario where lot of offline messages are being stored in mysql
database for a user B. As soon as destination (B) REGISTER with kamailio,
these messages will deliver to him. Now, kamailio will deliver all of these
messages that will cause user (B) device (softphone on mobile) stuck or
freezs for the moment messages are being delivered like bombardment from
kamailio.
Is there any way, where I can define batch size of messages to be delivered
after provided interval?
like batch_size=10 messages and batch_interval= 5sec, etc.
I want to avoid applying static limit to mysql query in module source code.
Please advise.
--
regards,
abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445
Hi,
can I with keepalive module ping with is_alive function endpoints connected
to kamailio?
Or this module can use only static destionations via modparam("keepalive",
"destination", "sip.provider.com")?
--
Aydar A. Kamalov
We are using kamailio 5.0 as an inbound dispatcher. Sometimes in testing we
see it take over half a second to relay a 200 OK. The logs show the new
dialog being creating for the 200 OK within .0001 seconds of receiving the
inbound 200 but it takes another .5 seconds for the relayed 200 to actually
hit the wire. Here is a screenshot showing the timing of the packets. The
box is an EC2 m3.medium under almost zero load (1-2 calls).
Is this normal? What should I be looking at?
Screenshot below: 10.0.23.34 is the kamailio box. Carrier IP obfuscated.
You can see the carrier resends the 200 OK because of the delay.
Hello everyone,
I have an error that I have not yet been able to solve and would like
the help of colleagues to indicate a correct path.
The problem that is occurring is that when the client disconnects the
call kamailio is not sending the BYE forward until arriving at the asterisk.
Both in the test scenario and in the production scenario the problem is
the same and the message I see in the capture is 404 Not here, msg this
coming from kamailio.
Production scenario.
PSTN <----------> Dialer --------->kamailio -----------> asterisk1
-----------> asterisk2
Test scenario.
sipp generated calls ------> kamailio -------> asterisk1
-------> asterisk2
When this occurs, the calls that are disconnected by the client are in a
"zombie" state in asterisk, and end up being terminated by timeout with
the following message in the asterisk CLI:
/[Apr 25 17:49:59] WARNING[2121]: chan_sip.c:4072 retrans_pkt:
Retransmission timeout reached on transmission 22-6073(a)10.110.7.242 for
seqno 1 (Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions//
//Packet timed out after 31999ms with no response/
In the sipp panel I see in the Retransmission column several
incrementing counters, as per the attachment.
If I take the kamailio from the move and point the sipp to only one of
the asterisk, the retransmissions do not happen and BYE follows normally.
My kamailio.cfg configuration file can be downloaded from this url:
https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=…
Thank you very much.
--
Hi All,
I've come accros a scenario where a proxy is sitting on a private
address range and inserts a record-route specifying the the private address.
This causes issues whereby the BYE to an INVITE attempts to relay to the
private address defined in record-route.
I was wondering, if we were inject a received and rport parameter into
the record-route header of the original invite, whould kamailio relay
the response (BYE) to the receive/rport destination instead of the uri
defined in the record-route header.
As an example, initial invite comes in with a record-route as follows:
Record-Route: <sip:172.17.0.2:5062;lr;ftag=b4551d29>
If we injected received and rport as follows:
Record-Route:
<sip:172.17.0.2:5062;lr;ftag=b4551d29;rport=33429;received=212.172.2.212>
and relayed the message to the B2B.
Then, I assume, when the B2B creates its BYE message, the Route header
should look like this:
Route:
<sip:172.17.0.2:5062;lr;ftag=b4551d29;rport=33429;received=212.172.2.212>
Once this hits the kamailio instance to relay to the last route header
as mentioned above, would it set $du to received:rport like it does with
Via headers?
Thanks
Hi all,
Just wanted to know what your opinions were on using DMQ modules over
database for things like dialog replication, registrations, etc...
Is DMQ the "new way to go"? I know that there lots of ways of doing things
with each having pros/cons... But I was wondering...
What does the community think on this topic?
Are you guys taking advantage of the DMQ modules or are you still relying
on database as much as possible? Maybe a combination of both?
Cheers,
Joel.
Hey All,
We are load testing Kamailio 4.2.3 and we are seeing the waiting within the transaction manager increasing. I’ve increased the children from 8 to 16 to 24, but we are still seeing a large amount of waiting. I don’t see an errors in the logs. Can anyone give me a clue on where to look.
05-02-18_03:07:38 PM
tm.stats
{
current: 390
waiting: 226
total: 2361
total_local: 0
replied_locally: 1406
6xx: 0
5xx: 0
4xx: 88
3xx: 0
2xx: 3180
created: 2361
freed: 1971
delayed_free: 0
}
Mack Hendricks / Head of Support / dOpenSource
web: http://dopensource.com <http://dopensource.com/>
support: +888-907-2085
dSIPRouter <http://dsiprouter.org/> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services
Phone sends Subscribe
Kamailio Responds 202 OK
Kamailio Responds with NOTIFY
Via: SIP/2.0/TCP 66.66.66.66:5051;branch=z9hG4bKfe3e.e4fedb84000000000000000000000000.0
To: <sip:30041@66.66.66.66 <mailto:30041@66.66.66.66>>;tag=48ae253ab7
From: <sip:47701@66.66.66.66 <mailto:47701@66.66.66.66>>;tag=00c39aabf419ddeab4250cd65af8ef90-6fb2
CSeq: 2 NOTIFY
Call-ID: 1493ef52c41b6c1a
Content-Length: 0
User-Agent: kamailio (5.1.2 (x86_64/linux))
Max-Forwards: 70
Event: dialog
Contact: <sip:192.168.1.30;transport=udp>
Subscription-State: terminated;reason=timeout
In that packet I see Subscription-state: terminated;reason=timeout
Im not sure what I am missing or misconfigured. Any help would be great. I’ve attached my config.
Thanks.
