I'm just starting to evaluate Kamailio to be used as a front end to a couple
of Asterisk Servers.
Currently I have an Adobe Flex application that interfaces with Asterisk via
AMI using a socket connection. I searched on how to interface with Kamailio
but wasn't able to find anything that helped me. I would appreciate it if
someone could point me to where I can find information on how this can be
done using Kamailio. For example I would like my Flex Application to connect
to Kamailio to know the status of the SIP channels.
Thanks for any help!!!
Carlos
daniel,
you wrote:
This feature should be useful in mobile networks, to avoid stacking lot
of invalid contacts which could result in lot of branches causing many
retransmissions.
why would a sip ua that runs in a mobile network device be configured to
use udp instead of tcp?
poor battery life seems to be a big problem in so called smart phones.
wouldn't this kind of active pinging make it even worse?
-- juha
Folks,
I'm working on a 5 node network in the following topology:
Host1 - Edge Proxy 1 --- Core Proxy - Edge Proxy 2 - Host2
The proxy machines contain SIP Server installations, the hosts contain user agent software. I am able to make a call end to end between the hosts, but when I look at the traffic in wireshark, the core proxy is actually just forwarding the messages at the IP-layer. I would like actual SIP routing to occur, where a host would send the INVITE to the edge, and the edge has to forward it to the core etc, with these messages actually being processed at the SIP layer for all proxy nodes. I've been trying to use the dynamic routing module, thought I'm not sure this is really right for what I need, simple static routing would likely work as well. Is it necessary for me to rewrite the route{} functions of a proxy script? I'm not sure if I'm approaching this problem the right way. Eventually I would like to have multiple core servers and be able to failover between them, but trying to keep it simple for now.
Thanks,
Mike
--------------------------------
Michael A. Kaplan
Senior Research Scientist
Applied Communication Sciences
732-699-3805 (office)
973-699-6775 (cell)
mkaplan(a)appcomsci.com
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[score: 0.0000]
1.6 URIBL_SBL Contains an URL listed in the SBL blocklist
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1.0 RDNS_DYNAMIC Delivered to internal network by host with
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Hey guys. I am very new to Kamailio. Basically, at this time, I want to generate a SIP message (from an SMS) within Kannel, and send this message to my Kamailio server, which will forward the SIP message to another SIP server. Therefore my Kamailio Sip server would act as a single user agent connecting to only one other SIP server. So what I am unsure of is:
1. Is there a Kamailio interface (such as an HTTP server) which Kannel can connect to in order for Kannel to send multiple messages to my Kamailio Server, which in turn would be forwarded as SIP instant messages to another SIP server?
2. Which Kamailio api can I use to formulate and send the SIP instant messages?
3. Which Kamailio callback function could be called when I receive acknowledgement responses to the SIP IMs I have sent?
4. Which Kamailio callback function could be called when I receive new SIP instant messages sent to me?
It would be great if someone could help point me in the right direction.
Many thanks.
Fred
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[109.100.26.103 listed in zen.spamhaus.org]
3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL
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[109.100.26.103 listed in bb.barracudacentral.org]
1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist
[URIs: euroinjobs.com]
1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist
[URIs: euroinjobs.com]
-1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1%
[score: 0.0000]
0.8 RDNS_NONE Delivered to internal network by a host with no rDNS
Hello all,
I'm using Kamailio (on a public IP) as a border proxy
that handles NAT traversal (detection and keep alives) for several
Freeswitch servers which have public IPs as well. I'm having a problem
record-routing SUBSCRIBEs and NOTIFYs for nated UACs. If the UAC is
behind NAT, I am adding a Record-Route to the initial SUBSCRIBE and
NOTIFY request. This works great until the UAC (in this case a Cisco
SPA509G) refreshes the subcription and sends the SUBSCRIBE directly to
the Freeswitch server. Since the request doesn't pass through the proxy,
Freeswitch then sends NOTIFYs directly to the UAC which only recieves
the first NOTIFY as the NAT pinhole is still open from the SUBSCRIBE.
The UAC did receive a NOTIFY validating the subscription, and continues
to refresh the subscription directly to the freeswitch server. The
completely breaks presence.
What is the correct way to ensure all
SUBSCRIBE/NOTIFYs pass through the proxy? I have found that I can force
this behavior by modifying the Contact header in the 202 Accepted to
point back to the proxy but I'm not sure if this is what I "should" be
doing.
Thanks for any assistance and please forgive my ignorance when
is comes to SIP presence!
Spencer
Hi All
I managed to figure out that the table had not been created in the database. So i deleted the install.lock file in the Siremis directory to run the install again. This time I clicked on all the boxes that were on step 2:
Create Siremis Database - this create database 'siremis'
Import Default Data - this imports initial data into database 'siremis'
Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules
Replace DB Config - this updates Siremis configuration file with the details to access databases
Then after it had finished I was able to login using the default admin/admin. Thanks for the reply Max greatly appreciated for the time taken to do so - Ivan - I didnt get chance to test what you had put however I will take a note of it for future reference.......hopefully I may be able to return the favor in the future if you guys need a hand.
Cheers guys
Dave
> From: sr-users-request(a)lists.sip-router.org
> Subject: sr-users Digest, Vol 87, Issue 64
> To: sr-users(a)lists.sip-router.org
> Date: Mon, 20 Aug 2012 16:17:22 +0200
>
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> Today's Topics:
>
> 1. Siremis login issue (dave mccann)
> 2. Re: Siremis login issue (Max M?hlbronner)
> 3. Re: [sr-dev] new feature: option to remove, contacts based on
> sip nat keepalives (Yufei Tao)
> 4. Re: Siremis login issue (Ivan Milivojevic)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 20 Aug 2012 12:51:57 +0100
> From: dave mccann <davemac2626(a)live.co.uk>
> Subject: [SR-Users] Siremis login issue
> To: <sr-users(a)lists.sip-router.org>
> Message-ID: <DUB117-W1040D3073CA62A3AE627CF787B90(a)phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> Hi All
>
> I have installed Kamailio with Siremis, Kamailio is working fine and starts and I have been through the initial setup for Siremis however when i go to login to the web management page it gives an error:
>
> "Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype,
> T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by,
> T0.create_time, T0.update_by, T0.update_time FROM `user` T0 WHERE
> T0.username='openser' and status='1' LIMIT 1. SQLSTATE[42S02]: Base
> table or view not found: 1146 Table 'siremis.user' doesn't exist"
>
> I have tried the default "admin/admin" login as mentioned in the install guide and have also tried the default passwords for Kamailio "openser - (with default password 'openserrw') and openserro - (with default password 'openserro'). I have also tried root/*password* and also siremis/siremis which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however none were successful and the same error appeared. I read that it was a permissions error however have set directory /var/www/siremis-3.2.1 to chmod 777 including all sub dirs and files.
>
> Can anyone please help as this is that last bit that I need to overcome :)
>
> Many thanks and best regards.
>
> Dave
>
>