Hey guys,
I don't seem to get BYE records from SER being sent to mysql, I get INVITE
and ACK. Without BYE I can't see who terminated the call, caller or callee,
and I can't determin the duration of the call.
Any ideas?
Thanks
Barry
oy u stop sending for me any stupid e-mails i don't wont them, if u will
send me another e-mail i will call to police for sending to me unwonted e-mails,
thanks
Hi,
I am trying to deploy a solution as shown below:
SER<------>Asterisk1
|
--->Asterisk2
|
--->Asterisk3
Calls come in to the SER machine. I want to distribute calls among the
asterisk machines(all with the same specifications). I initially tried
using DNS round robin. It works well but doesn not have builtin failover
mechanisms. So if a machine goes down calls will not be rerouted and
sent to other available machines.
I then tried using DNS SRV but, it seems, unfortunately SER does not
support failover support yet. Now I am not sure how to distribute load
amongst these machines. Any ideas suggestions will greatly be appreciated.
Regards,
Danish
Hi guys,
I'm working of configurations examples, how to get Ser and Asterisk
working together in peace ;)
What I wan to do is to have ser doing all the hard work, registering,
billing, proxying and help at nat stuff, and use asterisk to register
myself at my sip providers, as I couldn't found a way to do it with ser.
An idea how the things are or will be connected is...
[ UAC's ] -> SER ->
* / (Ser?) <-> [firewall full port redirect]
->
router
-> sip provider
192.168.20.x 192.168.20.1
192.168.20.10|192.168.30.10 192.168.30.1
192.168.30.x | 172.16.32.1 | 172.16.32.254 |external ip semi
fixed or fixed
I have all the stuff running fine if I use an IAX provider, problem
is... all my providers use sip :(
Does anyone has any clue to point me out ? Sip session goes fine but it
just get stuck in rtp nat and I can't understand very well why. Another
is the UAC's try to reach by itself the sip provider jupping over all
192.168.20.x/30.x thins goes fine if I use stun of course.
My question is... Is there any way to do the same in asterisk? Like use
a tun server or something, or using ser to help it to change any contact
or sip headers field? HOw is the best way to acomplished that?
Best Regards,
Hello, I have recently been put in charge of starting up a VoIP project
connecting voip trunk lines to business' existing PBX's. I have decided
to go all open source and have figured out asterisk pretty well. The
problem is with SER. There does not seem to be a lot of clear
documentation on asterisk interoperability. My idea is to use SER for
the bulk of the call routing and asterisk for voicemail, billing, and
other apps. All calls will be sip to me and then passed onto a VoIP
provider who will handle all Long distance and PSTN service. Does
anyone have any suggestions on setup, or any config files they could
email me please?
Thanks
Please give your help ... thanks ...
Hi SerUser!
Can somebody help me with this problem..
After I install and configure serweb as stated on configuration guide.. I
got a blank page
accessing admin and user_interface page.
On logs of my apache, this is the feedback ..
210.23.237.236 - - [21/Apr/2005:23:51:12 +0800] "GET
/serweb/user_interface/index.php?phplib_Session=4026e0eaf3aa336098b7e1d2926f49ff&phplib_Session=dc200ec3587d1985cf9aaf706d3b248c&phplib_Session=8d66aaa5f4418cb75126beda37c2acde&phplib_Session=14d85207dd2ec6439325db9f78bea914&phplib_Session=3a23bcfc1ecae59dcefead744c64aaf3&phplib_Session=6ff4d142925c0574d1d3614bcb679578&phplib_Session=5d83ca96f291e0310ab4777442a547f9&phplib_Session=8ac2999cee413bde7ef4769b8f73b860&phplib_Session=96c5b130e01b2db96d73bc050248183e&phplib_Session=9ae3f9631ab331d0c0299cf394e6f691&phplib_Session=ddb01b22fdc4d988a9093a37cf4ee00b&phplib_Session=d48993a3af2167a562d151f2cb285ce4&phplib_Session=2492b77a05e07417c38d229d30f58d9a&phplib_Session=a4edc5415a40e4a922c4dbf721f25dc0&phplib_Session=5d41839c67fbc842ac27f59a558e982a&phplib_Session=7b5b95a63d36ffdf9b412264bf941691&phplib_Session=507571c99dc312b6e36ea364de2644f4&phplib_Session=0992981f5a11f248824edbb95f5db9f3&phplib_Session=1f25b8f300e72b404a54ab4c2c4060e8&phplib_Session=870611806d28cda0d0021f8422454!
12a&ph
p
lib_Session=f5c422596455228e1749a081e1d9be1e&phplib_Session=2ecc3410c0ef13c4a66f12d51b02e7a3&phplib_Session=ff433b28f8596fe03518398fef080887&phplib_Session=aec18ca4299ec9009c236cde5d0bb733&phplib_Session=5937f583e214b20d67999da9947a9b7c&phplib_Session=854621f5353598bf6b6c9fd0f882766a&phplib_Session=0809e0fb3fddff21a14190e4240b814a&phplib_Session=22b80c053272e22c30c33a883182e014&phplib_Session=7c69db32533cd64b763462353c20d927&phplib_Session=c4eeba51a9991c5161269dafa9647d38&phplib_Session=472ca546c8ff6970ea9cae514940b86f&phplib_Session=c3c6871818a255d4eafe2d8a373228ef&phplib_Session=fded6f2f186c6d2dedd467a2b93f5c2a&phplib_Session=53033ca88aa7ec64b25d70f8ac424086&phplib_Session=62744ffaf3c7866e2f3324b1d6ca9f69&phplib_Session=d43a04d83057e27859e9a9db63782a4c&phplib_Session=bb309e7809c1341764212cab865e33ce&phplib_Session=002136a613c2bdee8d927d8a43be20ad&phplib_Session=82a678edfc54753212929c25f5fb5221&phplib_Session=2de93e7370b1de73041c6f4dc21a11df&phplib_Session=dfd10093b5f05cccd!
3178ad
f
cc3f8c62 HTTP/1.1" 302 -
Is there something I miss during the configuration ..
thanks,
ed
Hi!
I have the following scenario:
UA1 ----> SER1+NAT ----> Internet <----DNAT ROUTER <---- SER2 <----UA2
Both SER1+NAT and SER2 have 2 network interfaces and work with mhomed=1.
SER1 is on the same machine as the NAT, so SER1 listens on a public address.
SER2 is behind a router doing full DNAT to it, so it doesn't directly
listen in a public address.
Communication works in both ways, regardless of who starts the call!
This is the good thing.
The bad thing is, that when UA1 hangs up, the BYE request is sent to the
IP address of the outmost interface of SER2, which is behind the DNAT
router.
How do I make SER1 make the bye request go to the public interface of
the router?
I've tried changing the advertised_address and record_route_preset
parameters, and haven't achieved success yet!
Has anyone faced a similar problem, or can give me hand with this problem?
Thanx a lot!!!
Óscar
Hello,
I have some MC3810s that I want to use with SER over their PRI ports.
The problem with these is that IOS 12.3 doesn't seem to support SIP
registrar functions, so there is no way to get the DIDs to register with
SER.
Since these routers have static IP addresses, is it possible to have SER
authenticate based on IP address? I know I could just add a ton of lines
to my config file authorizing and routing them manually, but thoguht
someone might know a better way to save their location like the
registered users, and check the source IP vs the password field
(something like put the IP address in the password field).
I'm sure I'm far from the first that's ran into this, but couldn't find
much so thought I'd ask for ideas.
Thanks,
Brian
Hello,
I'm not sure if this could help, but i think the AVPOPS module can help you
to solve this problem. You can add or remove the "trusted" ip's from your
DB and let avpops do the work. So you don't need to modify your
configuration file. Check this link :
http://www.voice-system.ro/docs/avpops/
And check this example :
http://www.voice-system.ro/docs/avpops/ar01s08.html#ex_trusting
I think that could help you.
I'm in the way to try this too.
Good Luck
Regards,
Ricardo Martinez.-
> -----Mensaje original-----
> De: Brian McCrary [mailto:bmccrary@lwav.net]
> Enviado el: Jueves, 21 de Abril de 2005 11:54
> Para: serusers(a)lists.iptel.org
> Asunto: [Serusers] SER authorization based on IP address
>
>
> Hello,
>
> I have some MC3810s that I want to use with SER over their PRI ports.
> The problem with these is that IOS 12.3 doesn't seem to support SIP
> registrar functions, so there is no way to get the DIDs to
> register with
> SER.
>
> Since these routers have static IP addresses, is it possible
> to have SER
> authenticate based on IP address? I know I could just add a
> ton of lines
> to my config file authorizing and routing them manually, but thoguht
> someone might know a better way to save their location like the
> registered users, and check the source IP vs the password field
> (something like put the IP address in the password field).
>
> I'm sure I'm far from the first that's ran into this, but
> couldn't find
> much so thought I'd ask for ideas.
>
> Thanks,
>
> Brian
>
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>
