Hi Guys
I'm trying to get T38 working between our VoIP Provider and my epygi PBX.
I have the epygi registered with openSER when a phone call is received
from our provider openSER forwards the call to the epygi. When this unit
detects the fax tone it does a re invite. (the re invite has the SDP
indicating it want a T38)
This re invite never gets to our VoIP provider. OpenSER is blocking the
request.
The invite would seem to be hitting our loose routing
if (loose_route()) {
xlog("L_NOTICE", "Running loose_route logic");
if ((is_method("INVITE|REFER") && ! has_totag())) {
route(ROUTE_AUTHTEST);
if (retcode == FALSE)
return;
};
if (method=="INVITE") {
route(ROUTE_AUTHTEST);
if (retcode == FALSE)
return;
};
append_hf("P-hint: rr-enforced\r\n");
route(ROUTE_DEFAULTHANDLER);
return;
};
route[ROUTE_DEFAULTHANDLER] {
t_on_reply("REPLYROUTE");
if (!t_relay()) {
sl_reply_error();
};
}
Any ideas as to what I might have wrong ?
Keep in mind that we are using OpenSER as a part of VoIP service to a
small number of customers.
Thanks
Mike
Hi Friends,
How we can save Active Calls in openser mysql database?
Please advise us i need to display it on our billing system.
Regards,
www.Go4Calls.Com
VoIP Forums
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May you feel the Angels enfold you in their wings.
May you always find serenity in the simple things.
May the Light of Heaven shine upon your path,
and bring you to the completion of your work in Peace and Joy and Grace.
Wishing You All the Best this Christmas & a Happy new Year
Welcome Year 2008
Tuune Mobile
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Tel:012-382-5953
When I included a pv transformation in my cfg script to extract the expires value from a Contact I found that it would not work if the Contact string contained spaces. I does not seem to affect registration.
$(ct{param.value,expires})
The following patch in 1.3.0 works for me --
--- parser/parse_param.c-orig 2007-12-22 23:23:05.000000000 -0800
+++ parser/parse_param.c 2007-12-22 23:23:38.000000000 -0800
@@ -281,7 +281,7 @@
while(_s->len) {
switch(_s->s[0]) {
- case ' ':
+ //case ' ':
case '\t':
case '\r':
case '\n':
An example of a message that fails Contact expires transformation --
Session Initiation Protocol
Request-Line: REGISTER sip:proxy01.sipphone.com SIP/2.0
Method: REGISTER
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 192.168.1.4:5061;branch=z9hG4bK-1849b3b3
From: Gateway <sip:17476316719@proxy01.sipphone.com>;tag=a0bda46794e0291o1
To: Gateway <sip:1234567@proxy01.sipphone.com>
Call-ID: 6d0c06a5-8d11d464(a)192.168.1.4
CSeq: 51635 REGISTER
Max-Forwards: 70
Authorization: Digest username="1234567",realm="proxy01.sipphone.com",nonce="476e0e7866cafbf20434684456644a3d2022c85f",uri="sip:proxy01.sipphone.com",algorithm=MD5,response="57d8e4aea67f9bb45ced0394fe54d659"
Contact: Gateway <sip:1234567@192.168.1.4:5061>;expires=3600
User-Agent: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Folcks,
The problem
===========
supose that there is an established rtp connection between two endpoints
and the media proxy is in the middle doing the relay of the rtp streams,
let´s say
(MP = mediaproxy)
EP A <--------->[NAT with IP1]<-----------> MP <--------------------> EP B
EP A sends rtp to MP_IP:MP_PORT passing through the NAT box.
EP B sends rtp to MP_IP:MP_PORT without passing through a NAT box.
The MP know that the caller = NAT_IP1:NAT_PORT1, and the called =
EP_B_IP:EP_B_PORT
Now, supose that the NAT box change their PUBLIC IP from IP1 to IP2, so
this escenary
EP A <------->[NAT with IP1]<---------> MP <-------------> EP B
will change to this
EP A <------->[NAT with IP2]<---------> MP <--------------> EP B
so the MP should detect that change of IPs and continue relaying the rtp
streams but now to IP2:PORT2 instead of IP1:PORT1.
Well, that was the situation y have experienced.
The solution
============
To fix this, I was thinking this solution:
1) When the first rtp packet of a source arrives, save the SSRC field in
the MP.
- Save the SSRC of the caller.
- Save the SSRC of the called.
2) If arrives a rtp packet with unknown source IP but with the same SSRC
field of some of the two streams, updates the binding (with the new IP
detected) between the caller and the MP or between the called and the MP
according to the field SSRC previously saved.
Note: SSRC (RFC 3550 RTP), (from the rfc: "The SSRC identifier carried in
the RTP header and in various fields of RTCP packets is a random 32-bit
number that is required to be globally unique within an RTP session ")
What do you think about the solution?
Regards,
--
Gonzalo J. Sambucaro
Ingeniería de Software
Tel: +54-341-4230504
MSLC
gonzalo(a)mslc.com.ar
www.mslc.com.ar
Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica
Ciudad Universitaria Rosario UNR, CCT CONICET
Rosario - Santa Fé - Argentina
Hi,
I have following issue.
All my outgoing requests are forwarded through boundary proxy
by setting the destination before t_relay()
my problem is when there is another branch created.
This second branch is routed in the standard SIP way
directly to destination in request URI.
How can I force also branches to go through my
boundary proxy.
Can I somehow check in branch_route that this branch is not the first one
(not first message)
but the second and and change the destination and make t_relay()?
Thank You for any feedback
Regards
Tomasz
Dear all
anybody have done this job integration SER with Asterisk ?????
$ cat ~/satish/url.txt
http://www.linuxbug.org
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Hello,
I am using openser-1.1 and have the standard script
from onsip. I am using Mediaproxy for NAT traversal.
Most of my clients register for days with no issues. A
few clients after say a day of operation suddenly
start giving getting de-registered. I can see the ATA
or UAC trying to register to openser but openser gives
a reply back to the UAC saying "401 Unauthorized".
The UAC's which give this issue are behind standard
home grade wireless routers (netgear/linksys type).
Any idea why this may happen.
Thanks in advance
Roger
Hi all:
I know that the example uses REFER sip method and follow such process:
1. prepare two SIP softphone(eyebeam 1.5.16.1 ) , register to openser
2. user enter Caller and Callee's SIP Address
3. Web Server receive http request, then construct FIFO command,
using t_uac_dlg MI command, build Refer SIP message with From and
To header using user input
4. t_uac_dlg is invoked, send sip REFER message to Caller Sip softphone,
softphone return 481 Call Leg/Transaction Does Not Exist, so the call
can't be setup.
I have seen the notice"/Unfortunately, this example does not work. The
reason is use of REFER for third-party call-control has not been
standardized due to resistance of proponents of B2BUA use (which is
somewhat bloated and not always operational)". But I still wonder if
any softphone support this example?
My openser version: 1.3.0
Thanks.
Regards,
Chen Xueqin
/
I wish Merry Christmas and a Happy New Year 2008 to everybody helping with the project, from code to feedback and reports.
It was a full year of activities, with two major releases, lot of enhancements, fixes, and participations to events around the world. I think OpenSER fulfills most of the requirements its users need and it will keep being focused on robustness, performance, flexibility and new enhancements.
Looking forward to a 2008 of successes.
Cheers,
Daniel
