Hello:
I just received a request for what I am calling shared line appearance.
To me this means two IP phones register to SER with the same number.
When a call arrives for that number one party answers the call on
phone #1, places the call on hold then picks up the same call on
phone #2. I can think of a way to make this work with call park but I
do not think this is the seamless way our users want the process to
work. Has anyone had to address this issue?
Thanks,Steve
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
hi
following example:
uac1-->pbx
/
pstngw---->ser----->
\
uac2
uac1 is user1 with pstn headnumber 12345 extensions 0-100,
rpid=12345,alias=12345
if uac2 (or anybody from the pstn) calls 1234577 (or any other extension of
uac1), how can i forward only the extensions to uac1?
regards
raid
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Hello,
I have made some test with cisco ip phone & blind transfer and I have a
question.
Example:
A: a(a)toto.com
B: b(a)toto.com
C: c(a)toto.com
M: mobile(a)toto.com
A calls B but B forward the call to C, C answer the call and transfer the
call to a mobile phone M via a pstn gateway.
When C sends the REFER request to A we have something like this:
REFER.
From: b(a)toto.com
To: a(a)toto.com
Refer-To: mobile(a)toto.com
Referred-By: b(a)toto.com
And then when A sends the Invite to the gateways:
INVITE.
From: a(a)toto.com
To: mobile(a)toto.com
Referred-By: b(a)toto.com
But the call is transferred by C and not by B so the Referred-By: is not
correct. Is that an error from the cisco ip phone (I think not) and how can
I bill this call (I do the billing on the gateways).
Thanks.
Laurent
Hi,
in ser.cfg.sample I see that the answer related to subscribe message for unregistered user is "404 -Not found", this makes the subscription session end and so the UA doesn't send automatically other subscribe messages.
Is there any other possible answer that avoids this behavior?
Best regards, tex
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hi Ser-users,
[intro]
i'm sorry if my post is redundant but i really need to know!!
[problem]
I haven't root's privileges on a linux Fedora core 3 Server but i really would like install SER and Mediaproxy(for NATed client) on my server
i tried by
make prefix=/home/my_folder/ser_production
make prefix=/home/my_folder/ser_production modules
make prefix=/home/my_folder/ser_production install
but i've a lot of troubles (while compiling and while running ./ser o ./serctl)
[Question]
is it possible to install SER on my linux Box without root's permissions?
thanks in advance!!
Alex
Samuel Osorio Calvo wrote:
> Hi,
>
>
>>I'm not sure, but I think you can handle it this way. You can add an
>>alias sip:mobilenumber@gateway to the user, and an alias to itself.
>>Then, after lookup(alias) you will have 2 brachnes, the original one and
>>the mobilenumber@gateway. Then, lookup(location) will add the branches
>
>>from the location table.
>
> Unfortunately, I think it won't work because the first lookup (in aliases) will rewrite the first contact as Req-URI and the second binding will be added as an extra branch.
> When you make the second lookup (in location), only the Req-URI is checked in the location table (which is the one rewritten by the first lookup), loosing the branch obtained in the first lookup (aliases).
>
> Please, correct me if I'm wrong...
Yes, but if you have 2 aliases:
1. sip:a@a -> sip:a@a
2. sip:a@a -> sip:number@gateway
this should work - once I had misconfiured my aliases in this way and
behave like this. Nevertheless, using a permanent entry in the location
table is a much cleaner approach.
regards,
klaus
Hi serusers, another NAT issue from mine...
when I have two parts calling each other inside the same NAT (behind a router
with public IP different from ser's) I directly relay the call between the
two, without use of rtpproxy. The good thing is that audio works perfectly
both ways, the bad one is that SIP messages from caller reach ser but aren't
relayed to the callee. So ACKs and BYEs reach too many hops and if the caller
hangs up, the callee remains connected until he hangs up too.
On the other side (BYE from callee to caller) everything works fine.
By "ngrepping" on ser port I noticed this difference on the header of SIP BYE,
which from callee to caller is
BYE sip:caller@CALLER_PUBLIC_IP (=CALLEE_PUBLIC_IP)
and from caller to callee is
BYE sip:callee@SER_PUBLIC_IP
and ser keeps relaying to himself...
INVITE routing logic is something like this
if (loose_route()) {
if (has_totag() && method=="INVITE") {
if (nat_uac_test("19")) {
setflag(6);
force_rport();
fix_nated_contact();
};
force_rtp_proxy("l");
t_on_reply("1");
};
t_relay();
break;
}
if (method=="INVITE") {
record_route();
if (nat_uac_test(19)) {
fix_nated_contact();
force_rport();
setflag(6);
}
if ( !isflagset(6) || SAME_NAT) {
t_relay();
} else {
force_rtp_proxy();
t_on_reply("1");
t_relay();
}
}
If I ignore same NAT test and force the call thru rtpproxy, all works as it
should. I thought the problem was in fix_nated_contact and force_rport, but
removing the two lines does nothing.
any ideas? thanks in advance
--
Giovanni Balasso
giaso(a)yahoo.it
Hi,
>I'm not sure, but I think you can handle it this way. You can add an
>alias sip:mobilenumber@gateway to the user, and an alias to itself.
>Then, after lookup(alias) you will have 2 brachnes, the original one and
>the mobilenumber@gateway. Then, lookup(location) will add the branches
>from the location table.
Unfortunately, I think it won't work because the first lookup (in aliases) will rewrite the first contact as Req-URI and the second binding will be added as an extra branch.
When you make the second lookup (in location), only the Req-URI is checked in the location table (which is the one rewritten by the first lookup), loosing the branch obtained in the first lookup (aliases).
Please, correct me if I'm wrong...
Samuel.
Unclassified.
Hi
Have been trying to get get rpid , and cli working, dropped xlite to
test with grandstream just in case xlite didnt like it.
I have rpid=12345 in my subscriber db, against username 040600
In debug I get
save_rpid(): rpid value is '12345'
4(29312) authorize(): set string AVP 'rpid = 12345'
when I add
append_rpid_hf("<sip:",
">;party=calling;id-type=subscriber;screen=yes") to my ser.cfg just
before I t_relay, now in theory (someones :-)) this should add the
header, but using ngrep I dont see any headers added.
Iqbal
Hi
I am using this block in my t_reply route, and it seems that it from
here that the rpid isnt being added, whereas on the first INVITE I seem
to be adding it. However I cannot do what Ihave written below...
if (!search("^Content-Length:\ 0")) {
append_rpid_hf("<sip:",
">;party=calling;id-type=subscriber;screen=yes");
use_media_proxy();
};
Iqbal
