Dear All,
We are SER as Registrar server and sending REGISTER message using SIPP. Now,
after sending REGISTER message it is being received by the SER. The utility
/usr/local/sbin/serctl ul show correctly shows the contact details but it is
NOT writing in the text file.
The configuration file for SER is being attached for kind reference.
Any pointer will be highly appreciated.
Regards,
Ramachandran
Hi,
I am having trouble with my sip-phone recognising my domain name, and I am
wondering if I have set my DNS records correctly. If someone can see
something that is missing from my DNS record, I would really appreciate some
help. So far, I can register using my IP address, but I can't register using
sip.mydomain.com, which is what I would like. Any suggestions?
Many thanks,
Mike
$TTL 14400
@ 86400 IN SOA domain1.com. domain2.com. (
2006051301 ; serial, todays date+todays
86400 ; refresh, seconds
7200 ; retry, seconds
3600000 ; expire, seconds
86400 ) ; minimum, seconds
mydomain.com. 86400 IN NS ns1.mydomain.com.
mydomain.com. IN A 1.2.3.4
localhost.mydomain.com. IN A 127.0.0.1
mydomain.com. IN MX 0 mydomain.com.
_sip._udp IN SRV 0 0 5060 sip.mydomain.com.
mail IN CNAME mydomain.com.
www IN CNAME mydomain.com.
ftp IN A 1.2.3.4
ns1 IN A 1.2.3.4
sip IN A 1.2.3.4
Hi,
I was wondering on OpenSER and came to a Question: How can call routing to
SIP gateways be done based on their capacity? Lets say we have two SIP
gateways each capable of handling 60 cuncurrent calls. How could I tell
openser to forward the call number 61 to the next gateway? How would open
ser know if the first gateway has free resources (previuos calls ended), and
so on.
Thanks,
Hamid
Hi All,
Can you tell me how to configure SER as location server with dbtext mode?
If so, please send me the configuration details for the same ASAP.
Thanks for your co-operation.
Regards,
Ramachandran G
Hi Greger. On October 25, 2005, you wrote:
> At least one thing is for sure: I have now registered a two-digit number
> of people who are struggling with and proposing various solutions to load
> balancing/failover. We really need to find a solution soon, so all these
> bright people can spend their resources on tackling problems that will
> bring ser even further!! Such a solution should be a "best practice"
> that is *good enough*, and I would vote for simplicity. People who
> really need that top-performance/hardware are capable of tuning and
> fixing themselves (and maybe improve best-practice along the way), all
> the others need a simple, well-described setup.
> That is why I have been working with Andreas to try to mirror his
> efforts in an onsip.org setup that we will document and make
> available as soon as it is ready for prime time. I suggest that anybody
> who have opinions or suggestions put them forward so that everything will
> be taken into consideration.
I was wondering what progress has been made on this front. Obviously the
documentation isn't ready since it hasn't been released, but I'd be
interested in reading what's been done so far, and contributing my
knowledge wherever I can.
Cheers,
-- Nick
e: nick.hoffman(a)altcall.com
p: +61 7 5591 3588
f: +61 7 5591 6588
If you receive this email by mistake, please notify us and do not make any
use of the email. We do not waive any privilege, confidentiality or
copyright associated with it.
I download the xlite3.0 and testing with Asterisk server ,All the function is all right but video function cann't start to the caller ,Can anyones tell what's the back server needed? Is anyone know why?
Regard
ZhaoMin
----- Original Message -----
From: "Andrey Kouprianov" <andrey.kouprianov(a)gmail.com>
To: <serusers(a)iptel.org>
Sent: Friday, July 14, 2006 4:16 PM
Subject: Re: [Serusers] OT: xlite now with video and IM
> Yes, X-lite ver.3 now even supports presence and looks nicer. However,
> as far as i know, it has now only 2 concurrent lines, instead of 3.
>
> Andrey.
>
> On 7/14/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
>> Hi!
>>
>> I just found out that xten offers a new xlite, based on eyebeam. AFAIK
>> this is the first free available Windows SIP client with video and IM.
>> As IMO eyebeam is the best client for testing and debugging purposes
>> (because you can configure everything) I think this is interesting for
>> you too.
>>
>> regards
>> Klaus
>> _______________________________________________
>> Serusers mailing list
>> Serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
> _______________________________________________
> Serusers mailing list
> Serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
Hi all
Till now i have seen examples of rewriting with IP address and port
now i have another router from VoIP provider, where is give me sip username
and password with IP address and port
now i am planning to disconnect IP based authentication provider
and go with SIP provider, becoz of Quality and price
so how can i change the rewritehostport("provider IP:5060");
to my username/password:ip: port
can some one suggest me what is the best method to do
ram
Hi all!!
I'm using the t_relay_to() function from the tm module but I'm having
problems sending the message to the port that I want.
I have in my cenario a proxy server(port:5062) that relays the message to
another ser server that as the listening port 5063. Both ser servers are on
the same machine, but has you can see, in different ports.
I'm using t_relay_to("5063","10.10.12.4") for relaying the message, but
instead of going to port 5063 the message goes to the port 5060 where I have
another server.
What might be the problem? Can you help? tks a lot
Regards, Luis
Hi all
Till now i have seen examples of rewriting with IP address and port
now i have another router from VoIP provider, where is give me sip username
and password with IP address and port
now i am planning to disconnect IP based authentication provider
and go with SIP provider, becoz of Quality and price
so how can i change the rewritehostport("provider IP:5060");
to my username/password:ip: port
can some one suggest me what is the best method to do
ram
Hi everybody!
Have somebody seen the following error messages at syslog?
Jul 8 12:21:04 localhost /usr/sbin/ser[29329]: ERROR:
unforce_rtp_proxy: support for RTP proxy is disabled
Jul 8 12:21:04 localhost /usr/sbin/ser[29336]: ERROR: force_rtp_proxy2:
support for RTP proxy is disabled
Jul 8 12:21:04 localhost /usr/sbin/ser[29335]: Successethod - STOP
ACCOUNTING
Jul 8 12:21:04 localhost /usr/sbin/ser[29335]: ERROR:
send_rtpp_command: can't connect to RTP proxy
Jul 8 12:21:04 localhost /usr/sbin/ser[29335]: WARNING: rtpp_test:
can't get version of the RTP proxy
Jul 8 12:21:04 localhost /usr/sbin/ser[29335]: WARNING: rtpp_test:
support for RTP proxy has been disabled temporarily
Jul 8 12:21:04 localhost /usr/sbin/ser[29335]: ERROR:
unforce_rtp_proxy: support for RTP proxy is disabled
Jul 8 12:21:05 localhost /usr/sbin/ser[29341]: ERROR: force_rtp_proxy2:
support for RTP proxy is disabled
Jul 8 12:21:05 localhost /usr/sbin/ser[29329]: ERROR: force_rtp_proxy2:
support for RTP proxy is disabled
Are this normal?
Regards
Alberto Cruz
