I tried it as well. The GUI appears but when it tries to make a connection
with it's sever looks like it hangs up...
> I would not bet on it too much -- it does not seem to be actively
> maintained and the last version I tried was not exactly meeting
> my expectations.
>
> -jiri
>
> At 10:13 PM 4/23/2004, Sergio Diaz wrote:
>
>>Anybody work with SerAdmin (xten) last version and Ser 0.8.12 ?
>>
>>I try to install but not make to work....
>>
>>Any Ideas ?
>>
>>Regards!
>>
>>--
>>Sergio DÃaz Escobar. CCNA
>><sdiaz at comnet dot net dot mx>
>>Network Engineer - Information Technology
>>COMNET, S.A. DE C.V.
>>Patriotismo 889 7o. Piso Col. Mixcoac
>>Mexico, 039100, D.F.
>>(52) 55 12539230 <> 121 Fax. (52) 55 12539240
>>
>>
>>
>>Computers are unreliable, but humans are even more unreliable.
>>Any system which depends on human reliability is unreliable.
>> Murphy`s Law.
>>
>>
>>
>>___________________________________________________________________
>>
>>This email may be confidential and/or privileged. Only the intended
>>recipient may access or use it. Any dissemination, distribution or
>>copying of this email is strictly prohibited. If you are not the
>>intended recipient please notify us immediately by return email and
>>then erase the email.
>>
>>We use virus scanning software but exclude all liability for viruses
>>or similar in any attachment or message...,..,..,.
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>
> --
> Jiri Kuthan http://iptel.org/~jiri/
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Regards,
Lakmal
Lankacom Services (Pvt) Ltd.
65C, Dharmapala Mawatha,
Colombo 07.
Sri Lanka.
Tel: +94-1-437545
www.lankacom.net
Hi all,
It is expected to be cisco solution for sip<->H.323 in
2004
http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_qanda_item09…
Arnd, could you explain, why you think sipquest.com
sucks?
Thanks.
Esteban D. Benavides wrote:
> I'm looking for a gateway for SER to translate and
work between SIP and
> H.323. Is there any software for this...
No Open-Source solution available except Asterisk. It
is said that asterisk
would be capable of this but i havent seen it at work
or any example config.
Dont try "sipquest.com" products cause the company IMO
sucks.
If you only need H323 -> sip-proxy -> sipua
communication you can
use siph323csgw which is part of Vovidas "Vocal"
distribution. You
cant use siph323csgw for for SIP <> H323 GK/GW
communications.
best regards,
:: Arnd ::
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
__________________________________
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Hi all,
I started using SER a few weeks back. It's working
pretty good, but i am having some issues with audio.
This is the scenario:
softphone <-> NAT <-> SER/Public IP <-> NAT <->
softphone
I have a softphone in our local network trying to call
another softphone in another network. This works
perfectly, I am having no issues. But when I try to
call a softphone in the same local network, the
connection establishes, softphone says audio active,
but there is no sound. I think this is because of NAT,
but how can i make it work for phones connected in the
same network? Can anyone help me in figuring out the
problem, or any pointers or links from where i can
find some useful information? Any help on this regard
would be greately appreciated.
Regards, Girish
__________________________________
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Yes, it would be good if you captured all the SIP messages and send a error
description to serusers(a)lists.iptel.org. I am afraid, this is not really a isdngw
problem, but more a SER routing problem.
Uli.
On Friday 23 April 2004 13:06, Ulrich Holeschak wrote:
> Hello again,
> would it help if i create an log file with etherreal? I expect it's getting
> very big, but it contains more information. I expect, that i can only log
> on the windows side, so if there are telegrams going to the wrong
> destionation you can't see them ...
>
> Ulrich
>
> -----Ursprüngliche Nachricht-----
> Von: "Ulrich Abend" <ullstar(a)iptel.org>
> An: "Ulrich Holeschak" <ulrich(a)holeschak.de>
> Gesendet: Freitag, 23. April 2004 11:57
> Betreff: Re: [Sems] current cvs ser version and isdngw
>
> > Hi Ulrich,
> >
> > sorry for asking for the logfiles again, my hosting provider delayed your
>
> mail
>
> > from 14:01. I received it yesterday at 23:00 :-((
> >
> > Anyway, now after I reviewed the complete log files, I still cannot find
> > a reason for the strange behaviour.
> >
> > Please tell me again, what exactly happens, from your last log files I
>
> see,
>
> > that the connection is established for ~30sec. What causes your Messenger
>
> to
>
> > send BYE then?
> >
> > Apr 22 23:24:46 router ser[29099]: method: <BYE>
> >
> > Do you manually stop the connection? Is Audio working allright while the
> > 30sec?
> >
> > What version of Messenger do you use? There have been some problems with
>
> the
>
> > standards compliance (esp. version 5.0, recommended version is 4.7)...
> > Do you have any other SIP client, you could use for testing? E.g. kphone?
> >
> > Uli.
> >
Hello Laurent,
Tuesday, April 20, 2004, 6:21:07 PM, you wrote:
LB> I think i follow the instructions ... but should i "route" ( so modify
LB> the ser.cfg file) some messages to my gateway to use the answering
LB> machine or services provided by my gateway ...
LB> And what about autocreatepeer=yes?
surely later you will have to route in ser.cfg for outbound calls, checking first for
user auth and grp assignement.
Using the extention method you can direct sip addresses to services in
asterisk like voicemail or voice prompts.
that's my ser routing for outside calls
record_route();
if (uri=~"sip:0[0-9]+@"){
if (!proxy_authorize("mydomain.com", "subscriber")) {proxy_challenge("mydomain.com", "0");sl_send_reply("403", "That's not your home");break;}; #fine proxy challenge
if (!is_user_in("credentials", "local")){sl_send_reply("403", "No permission for local calls");break;}; #fine invite
rewritehostport("sip.mydomain.com:5090");
t_relay();
break;
}; #fine if uri sip:0
inside asterisk calls forwarded by ser are treated by this extention
exten => _0.,1,Dial,Zap/g1/${EXTEN:1}|45|r
exten => _0.,2,Congestion
its important to notice that you have to block port 5090 for incoming
ip requests ....
Only ser will be allowed to talk to asterisk and forward calls, after
user checking.
Hope it helps
LB> thx
LB> Alessio Focardi wrote:
>>Hello Laurent,
>>
>>Tuesday, April 20, 2004, 4:45:41 PM, you wrote:
>>
>>LB> But my clients should register on SER or on Asterisk?
>>
>>On ser, then you will need to protect asterisk from unallowed pstn
>>call, but that will come later on.
>>
>>
>>LB> thx
>>
>>LB> Alessio Focardi wrote:
>>
>>
>>
>>>>Hello Laurent,
>>>>
>>>>Tuesday, April 20, 2004, 1:50:37 PM, you wrote:
>>>>
>>>>LB> In fact, my ser installation works fine...
>>>>LB> I can pass call through asterisk in standalone...
>>>>LB> The problem is to interconnect the 2, to register ( i don't know if it's
>>>>LB> a right solution) the sipphones on SER and to go outside thanks to
>>>>LB> asterisk...
>>>>
>>>>make asterisk use port 5090 for sip, then as a first step make
>>>>asterisk register in ser as an extention.
>>>>
>>>>you can do this in asterisk's sip.conf
>>>>
>>>>example
>>>>
>>>>register => 10:password@sip.yourdomanin.com/10
>>>>
>>>>this tells asterisk to register extention 10 as address 109(a)yourdomain.com
>>>>
>>>>dial 10 with a sip phone and you are in asterisk ... note that you
>>>>should have an extention 10 defined, or it will not work.
>>>>
>>>>B> Thx
>>>>
>>>>LB> Alessio Focardi wrote:
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>>Hello Laurent,
>>>>>>
>>>>>>What you want to accomplish could be done, my advice is to setup a
>>>>>>working installation of ser then you will continue with asterisk.
>>>>>>
>>>>>>At first make a simple installation of ser (no auth, no db maybe) and make
>>>>>>your phones call each other.
>>>>>>
>>>>>>If you encounter specific problems and you want to have some help this
>>>>>>is the right place.
>>>>>>
>>>>>>Good luck !
>>>>>>
>>>>>>
>>>>>>
>>>>>>Tuesday, April 20, 2004, 1:29:41 PM, you wrote:
>>>>>>
>>>>>>LB> Hi,
>>>>>>LB> I want to use SER as a sip Proxy and asterisk as a gateway to the PSTN
>>>>>>LB> network ...
>>>>>>LB> My sipphones are BudgetTone101 and i'm having trouble trying configure
>>>>>>LB> them....
>>>>>>LB> Indeed, i don't know if they should register on SER or not... I don't
>>>>>>LB> know what kind of sip messages should be passed to my machine running
>>>>>>LB> asterisk.
>>>>>>LB> I don't know what must be in ser.cfg ( if you've an example it could
>>>>>>LB> help me a lot...)...
>>>>>>LB> I wasn't able to find documentations about using Ser and Asterisk in
>>>>>>LB> this configuration ( messages in the archives are not explicit
>>>>>>LB> enough....) , so if you've a pointer or so....
>>>>>>
>>>>>>
>>>>>>LB> Help...
>>>>>>
>>>>>>LB> thx,
>>>>>>LB> Laurent
>>>>>>
>>>>>>LB> _______________________________________________
>>>>>>LB> Serusers mailing list
>>>>>>LB> serusers(a)lists.iptel.org
>>>>>>LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>LB> _______________________________________________
>>>>LB> Serusers mailing list
>>>>LB> serusers(a)lists.iptel.org
>>>>LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>
>>
>>
>>
>>
--
Best regards,
Alessio mailto:afoc@interconnessioni.it
Hi,
does anyone have a hint for a good windows software for Marcello including the
g723.1 codec?
Thanks,
Uli.
----- Weitergeleitete Nachricht von Marcello Lupo <lupo(a)itspecialist.it> -----
Datum: Wed, 21 Apr 2004 23:46:35 +0200
Von: Marcello Lupo <lupo(a)itspecialist.it>
Antwort an: Marcello Lupo <lupo(a)itspecialist.it>
Betreff: SIP Client g723.1
An: ullstar(a)iptel.org
Hi Ulrich,
i hope you remember me...
I hope all is good with you.
We are using very happily your SER in our offices in production environment
right now. Thanks a lot for all you past help.
I have a question, do you know any very good client SIP Phone software running
on Windows that support the G723.1 codec?? I think that it will be almost
sure a commercial product but it doesn't matter.
Thank you again,
Bye,
MArcello
----- Ende der weitergeleiteten Nachricht -----
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Is the latest stable version of the the rtpproxy still v20040105 ?
I am struggling to compile it on RedHat 9 with make-3.79.1-17
I downloaded sorce files using cvs co -r v20040105 rtpproxy
when I use the make install command the following error message is returned.
"Makefile:33: *** missing separator. Stop.
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