25 Jul
2007
25 Jul
'07
7:30 a.m.
On 7/25/07, Weiter Leiter <bp4mls(a)googlemail.com> wrote:
--
"C is a language that combines all the elegance and power of assembly
language with all the readability and maintainability of assembly language."
Vlad,
SER is only handling signaling, SIP. It is unable to handle "voice packages"
- RTP in this case - by itself. That's why two other - equivalent - helping
applications exist: rtpproxy and mediaproxy.
So you are saying that I should also install these other two applications (
are they applications ? excuse me , but i'm not familiar with this and I dit
not quite undestood what you ment to say)
So, in your scenario with SER is the session established OK? In other words,
is the call established and lasts on both phones for more than 30s (or
more), or less than that: the callee picks up, but this is never perceived
at caller?
No the session is not established with SER (it gives me busy tone) ;
without SER (from the AS directly to the HT), the phone rings more than 30
sec , but when it is picked-up ... nothing.
And this is my problem ,that with SER the phone doesn't ring and I do not
know how to make this happen from the ser config file.
If it helps , here is the dial-peer from the AS router to send the packages
to SER:
dial-peer voice xxxxx voip
description tech support
huntstop
destination-pattern .%00111111T
redirect ip2ip
voice-class codec 11
session protocol sipv2
session target ipv4:193.226.xxx.xxx ----> the ip of the computer with
SER installed
dtmf-relay sip-notify
-I'm sure that this dial-peer is corect because , as i said before , it
worked with a hardware SipServer.
As last resort, do a network sniffing on the SER box and send it over.
This will take some time , because I will have to install Ethereal on linux.
On 7/25/07, Vlad Costea <vlad.costea(a)interpoint.ro> wrote:
No , they are on different networks, and NAT has nothing to do with it
because there is no router to do so between them, only the default gateways
( both HT and AS have public ip adresses) . As i said in the first mail , my
only problem is the config of SER to receive the voice packages from one IP
adress and send them to another IP adress and only that ( something like :
listen on : 193.226.xxx.xxx,5060 , send to: 193.230.xxx.xxx ).
SER is only handling signaling, SIP. It is unable to handle "voice
packages" - RTP in this case - by itself. That's why two other - equivalent
- helping applications exist: rtpproxy and mediaproxy.
Actually I'm not even sure that SER cand replace the hardware version of a
Sip Server . The path I described
(phone->AS5350->SipServer->HandyTone->phone) , works on an request/reply
system and the codec negotiation is made after the reply from the HT;
if the reply message is not transmited on the same path there is no codec
negotiation , there-for no voice.
So, in your scenario with SER is the session established OK? In other
words, is the call established and lasts on both phones for more than 30s
(or more), or less than that: the callee picks up, but this is never
perceived at caller?
Anyway , thanks for trying to help me. Unfortunately, day after day , I'm
begining to think that there is no solution for
my problem.
From my experience, unless running SER, no other (hardware) box can do as
much as SER&friends can. You just need to clearly say what hurts.
As last resort, do a network sniffing on the SER box and send it over.
WL.
On 7/25/07, Weiter Leiter <bp4mls(a)googlemail.com> wrote:
Are the AS and HT both part of the same network?
Otherwise, most comon diagnosis for your symtom is NAT and you might
find some inspiration as instructed below, or any other NAT support resource
you find on www.iptel.org
ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.cfg
ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-mediaproxy.cfg
On 7/24/07, Vlad Costea < vlad.costea(a)interpoint.ro> wrote:
Hello.
I have just installed SER on RedHat 9 Linux with the purpose to
route voice packages between a PSTN Gateway AS5350 and a Grandstream
HandyTone 286.Let me tell you how it should work: I have assigned a
phone number, that when it is picked-up, the IVR from the AS5350 router
respods and after the press of a key it sends the packages to the HandyTone
and from there to a normal phone; So far so good because it partialy works,
that means that the phone rings butt there is no voice. I have managed to
solve this problem using a hardware Sip Server between the two devices ( in
this way both HT286 and AS5350 act as clients), but this is not possible any
more because it was not my server; so i have tried with a software solution
and I've installed SER. As I said before , I haven't managed to configure
it to work as I wish and this is why I'am asking for your help; it would
realy help me if you can provide a ser.cfg example that would do
just the routing part (no ack no authentification , no mysql,...., just
routing).
Please excuse my bad english.
Thank you very much !
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--
"C is a language that combines all the elegance and power of assembly
language with all the readability and maintainability of assembly language."