Hello everyone,
At first I am sorry that this is slightly off topic.
I am doing some research for my company (which does not want to be
named therefore the gmail. I am sorry for that) which carrier grade
media-Gateways are used by others to connect their SER (or any other
SIP-Proxy, - Router, -PBX, etc.) to the PSTN.
We are currently looking for something that can be connected via SS7
in Europe, scales good (i.e. starting with 4-E1s, but with no limits
to add more) and supports
* SIP ;)
* G711, G723 and G729 A/B Codecs
* Echo Cancellation
* VAD
* CNG
* T.38
* Modem Bypass
* DTMF-Transmission using RFC2833 or In band (for G711) and
preferably even SIP-INFO
I would love to hear which system you use/prefer and what your
experiences are ( i.e. would you buy the system you use a second time)
and if I forgot something in my requirements that in your eyes is
crucial. If you don't want to/can name your company that's fine with
me (as I can't either), I just need some opinions.
Bye,
K
Hello everyone,
At first I am sorry that this is slightly off topic.
I am doing some research for my company (which does not want to be
named therefore the gmail. I am sorry for that) which carrier grade
media-Gateways are used by others to connect their SER (or any other
SIP-Proxy, - Router, -PBX, etc.) to the PSTN.
We are currently looking for something that can be connected via SS7
in Europe, scales good (i.e. starting with 4-E1s, but with no limits
to add more) and supports
* SIP ;)
* G711, G723 and G729 A/B Codecs
* Echo Cancellation
* VAD
* CNG
* T.38
* Modem Bypass
* DTMF-Transmission using RFC2833 or In band (for G711) and
preferably even SIP-INFO
I would love to hear which system you use/prefer and what your
experiences are ( i.e. would you buy the system you use a second time)
and if I forgot something in my requirements that in your eyes is
crucial. If you don't want to/can name your company that's fine with
me (as I can't either), I just need some opinions.
Bye,
K
hi i think is got a unique problem. i have a phone system that until the
final software package is developed will only allow a user name but no pass
and i need to make sure that the server isn't taken for advantage so only a
few very long and complex user names will be used.
i tried adding a user then going into the mysql and deleting the pass but it
seems ser doesn't even get that far if the client doesn't send a pass.
so to recap i need a user without a pass but only specified user names to
work.
also some guy hinted about gateways though I'm not sure he knew what he was
on about, so if any of you know that would be good and if their are any
documents in ti as well that you could point me to that would be good also.
thanks
Alex Wood
independent contractor
Hi all,
I am using presently ser-0.9.6 and have uac with in the ser. whenever i run
my application i get this warning
Warning: run_failure_handlers: no UAC support (0, 0)
I am not much clear about this warning . can someone give me an idea about
this.
Thanks&Regards
A.Suresh
I thought that fax over VoIP was unreliable in general? Even with G711,
doesn't the packetization break it? Any hints about this are greatly
appreciated.
---Mike
> Hello,
>
> On 04/04/06 11:11, Marc Haisenko wrote:
> > On Tuesday 04 April 2006 09:49, Christoph Fürstaller wrote:
> >> Hi,
> >>
> >> Is there anyone who uses fax over openser? Or can say something about
> >> the reliability of fax over openser?
> >>
> >> chris...
> >
> > We're not actively using it but we tested it. It works :-)
>
> what i can add here is to pay attention of managing re-INVITEs, in the
> case you have NATs around. Sending fax uses re-INVITEs to switch to fax
> data transmission.
>
> Cheers,
> Daniel
>
> > C'ya,
> > Marc
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
-------------------------------------------------------
Hello,
i have following behavior:
when a client is detected behind nat during REGISTER, and i do "fix
nated register" i have the public ip address in my location table.
but now, when there is an INVITE sent to this user and i append
branches, i can see that all invites are sent to the client ip address,
that makes no sense.
sample config:
route[1]{
.......................
append_branch();
avp_pushto("$ruri/username","s:msisdn"); ## ex:
0676123456(a)myproxy.com
rewritehostportport("10.0.0.1:5060"); ## my gateway
route(2);
}
route[2]{
if(!t_relay()){
sl_reply_error();
}
exit;
}
with ngrep i can see
Initial INVITE to this uri
trying...................
and the the proxy sends:
INVITE sip:initialuri@myproxy.com -> to the ip address that is in the
location table (in received field)
INVITE sip:0676123456@10.0.0.1:5060 -> also to the ip address that is in
the location table in the received field
if i make this tests without clients behind nat, all is working fine and
one invite is sent to the client and one to the gateway.
any ideas ?
best regards,
Andreas Matzel
Hi All,
I am facing a problem when terminating calls to Voice Master.
The exact problem is :
Voice Master is configured for Pin prefix Authentication.
I add the gateway prefix in my openser.cfg using prefix("123#")
But the Voice Master is rejecting the calls with error 488 Not acceptable
here.
User dialing number and openser adding prefix to it.
But when I send the calls from the userend itself with prefix ie. User is
dialing 123#<number> then the calls are going fine. What comes to my mind is
that Voice Master is somehow reading the to_uri being sent. Can I somehow
add prefix to the to_uri as well ?
This is the ngrep trace of the rejected call :
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060
INVITE sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest
username="test",realm="XXX.XXX.XXX.XX6",nonce="4433562d730464572c23207912e46
d50281d2136",response="59311eb34dd7ce8b3550064960af1508",uri="sip:1408526400
0(a)XXX.XXX.XXX.XX6".
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060
SIP/2.0 100 Trying.
From: <sip:test@XXX.XXX.XXX.XX2:5060>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX2:5060>;tag=669a9b6321e12fb3fd8a82d7.
CSeq: 38322 INVITE.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6:5060;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
Content-Length: 0.
.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060
SIP/2.0 488 Not Acceptable Here.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: "test" <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest username="test", realm="XXX.XXX.XXX.XX6",
nonce="4433562d730464572c23207912e46d50281d2136",
response="59311eb34dd7ce8b3550064960af1508",
uri="sip:14085264000@XXX.XXX.XXX.XX6".
Max-Forwards: 68.
Content-Type: application/sdp.
User-agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060
ACK sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
CSeq: 38322 ACK.
User-Agent: OpenSer (1.0.1 (i386/linux)).
Content-Length: 0.
.
exit
5 received, 0 dropped
Hi all,
I am trying to implement the call forwarding feature in openser. The forwarding part is fine. But I have the follwing scenario:
Whenever a user needs to change his forwarding number, he should be able to do so from his end device only.
For eg: He first presses 86 and then the 10 digit number to be forwarded. Here the 86 has to be stripped and the 10 digit number should be inserted into the call forward value column of the preferences table.
Also if he wants to remove the forwarded number, he can do so by pressing say for eg: 87. on receiving this number openser should delete that value from the
table.
Is this possible. I tried to do the following, but somehow it does not change the
value.
if(uri=~"^sip:86[0-9]*@") {
if(avp_db_load("$from/username", "s:callfwd")) { #check if call-fwd feature is enabled for the user
log(1,"AVP condition returned true");
strip(2);
avp_write("$ruri", "s:callfwd");
avp_print();
log(1,"AVP written");
sl_send_reply("200", "OK");
exit;
};
};
Is avp_write the proper method or I guess avp_db_store can also help me. The avp_print() function also does not show me anything in the log. Are there any
logical mistakes or I have mis-interpreted the syntax of avpops functions.
Please help me in thsi regard.
Thanks a lot in advance.
Jayesh.
---------------------------------
Jiyo cricket on Yahoo! India cricket
Yahoo! Messenger Mobile Stay in touch with your buddies all the time.
Hi
Somebody knows a reseller billing system for Openser ?
Thanks in advance.
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
Servidores BSD, Solaris y Linux
Soporte técnico ISPs
Jabber ID: rpereyra(a)lugmen.org.ar
For reliable and professional DNS, use DNS Made Easy!
http://www.dnsmadeeasy.com/u/14989
Hi,
i'm new in OpenSER and i like know, how i can hang up a current call?, i did
try it via fifo file, but i haven't found any command to do it.
Anybody can help me?
Regards.
--
Manuel A. Rubio "Bombadil"
Usuario de GNU/Linux #323628 acorde a http://counter.li.org/
GPG ID 1C84979D ftp://bosqueviejo.net/pub/bombadil.asc
Técnico en Admin. Sistemas Informáticos
