Hi Zoltán,
could you post the sip traffic (inbound and outbound) ? in what route
are you using the uac_restore_from() function?
anyhow, my advise is to upgrade to 1.2 version as in ~ 2 weeks the 1.3
will be release and 1.1 will become unsupported version.
regards,
bogdan
Zoltán Kovács wrote:
I found out that the uac_replace_from() function has a
weird behaviour
on my system. The same result occurs for all the three
modparam("uac","from_restore_mode",...) settings ("auto",
"none",
"manual").
1) I cannot change the display or the URI part in FROM header.
Whatever I do I'll get no change.
2) If a inbound hidden call is coming from an inbound trunk (#31#...)
then uac_replace_from("XXX", "sip:YYY") unhides the display and the
URI in FROM header --- however the displayed phone number will be
independent from XXX or YYY, it will be the original phone number of
the caller (XXX or YYY may be omitted, too). The text "sip:" cannot be
omitted.
I don't really have an idea what is going on here. Could you please
help? I'm still using openser-1.1.0.
Thank you in advance. Best regards, Zoltan
2007/11/27, Zoltán Kovács < kovzol(a)gmail.com <mailto:kovzol@gmail.com>>:
Dear Colleagues,
first of all, thank you for the great software which is a very
nice and convenient way to build up a stable VoIP telephone
system! We already have a working system based on OpenSER and
Asterisk.
After many experiments I finally raise this question to you
because I cannot figure out the solution. The problem described on
http://www.openser.org/pipermail/users/2006-January/002747.html
<http://www.openser.org/pipermail/users/2006-January/002747.html>
seems to be very similar to mine.
I am using openser-1.1.0-9etch1. I tried to replace the FROM
variable as follows:
loadmodule "uac.so"
modparam("uac","from_restore_mode","auto")
...
route
{
...
avp_write("$fU", "$avp(s:display)");
avp_write("$fU", "$avp(s:dispuri)");
avp_subst("$avp(s:display)", "/^36(.*)/06\1/");
avp_subst("$avp(s:dispuri)", "/^36(.*)/sip:06\1@datanet.hu/");
uac_replace_from("$avp(s:display)", "$avp(s:dispuri)");
...
}
Unfortunately nothing seems to happen: no change is made on
several VoIP telephones, including X-Lite as well. loose_route()
and record_route() were also used in my configuration. Could you
please help me what I am missing here? If needed, I can also send
a full config file.
Thank you in advance,
Zoltan Kovacs
sysadmin
--
matek.hu <http://matek.hu> --- mert matekból érettségizni mindenkinek
kell...
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