Hello all,
At this moment i'm using the full e164 numbers as account's in ser, that
is for every number someone wants delivered to them they would have to
register it using the account details.
This however poses a problem with a 100 bock of numbers for example
while a device can only register a few of them so what i would like to
do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way
the devices only has to register once and recieves the number dialed in
the incomming INVITE. If i use aliases to alias numbers to 1 account
the account allways recieves it's own registered contact instead of the
number dialed, is there any way to support number blocks in this way?
For example:
Device registeres as device_1@sipserver with contact device_1@device_ip:port
SER gets invite for number in the block of device_1 and sends the invite
as dialed_number@device_ip:port instead of device_1@device_ip:port
Kind reagards,
Erik Versaevel
I was testing with scripts of serr one receives calls of asteriskk and
the other can make calls to asterisk? I see that metodo INVITES in one
of scripts is repeated twice, in any case podrian to give a serious
guide to me of as script in form generates .
my idea is to cause that being can receive and send calls to asterisk.
thanks
Guys, do any of you know if there is a way in SER to dynamically set up
aliases using radius? I'm trying to achieve similar results but with
aliases.
Thank you.
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Chris St Denis
Sent: Wednesday, August 24, 2005 1:26 PM
To: 'Erik Versaevel'; 'SER Users'
Subject: RE: [Serusers] Routing multiple numbers to 1 account
I am doing this by pulling the translation out of a database using
radius
AVP (could be done direct to database too) and sticking it as the
destination.
It looks like this
# Try to lookup the number directly registered.
if(!lookup("ser_location")) {
#Number isn't found, so try to do an avp translation on
it
avp_pushto("$ruri/username", "s:callee_vdid");
if(!lookup("ser_location")) { #Look up the new value
#Still can't find the user. Give up.
sl_send_reply("404", "I can't find 'em");
break;
}
}
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Erik Versaevel
Sent: Wednesday, August 24, 2005 6:49 AM
To: SER Users
Subject: [Serusers] Routing multiple numbers to 1 account
Hello all,
At this moment i'm using the full e164 numbers as account's in ser, that
is for every number someone wants delivered to them they would have to
register it using the account details.
This however poses a problem with a 100 bock of numbers for example
while a device can only register a few of them so what i would like to
do is mimic the asterisk like dial(SIP/number@sipppeer) in SER. That way
the devices only has to register once and recieves the number dialed in
the incomming INVITE. If i use aliases to alias numbers to 1 account
the account allways recieves it's own registered contact instead of the
number dialed, is there any way to support number blocks in this way?
For example:
Device registeres as device_1@sipserver with contact
device_1@device_ip:port
SER gets invite for number in the block of device_1 and sends the invite
as dialed_number@device_ip:port instead of device_1@device_ip:port
Kind reagards,
Erik Versaevel
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_______________________________________________
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serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Is there any available api for java on VoIp.
- javabuddy.
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Hi
If I have a inbound call frompstn which hits SER, and then the call is
passed to asterisk, the receiver then decides to transfer the call, can
this be controlled within asterisk/ser, without the PSTN gateway being
involved, I was thiking that although REFER messages are sent, couldnt
SER take care of this and just transfer the call to a new sip contact.
The problem I have got is that I have a upstream gateway which doesnt
support REFER :-)
Iqbal
Hello,
I try debugging and get this error
"found match with different intervals (20 != 30)"
Is it possible to change the p-time into SER from 20 to something else like 30 or 40 ?
Thanks,
frank
---------------------------------
Start your day with Yahoo! - make it your home page
I am having a problem with incoming calls where ACK messages dont seem to
be getting (loose) routed.
Here is the relevant part of my config
======================
route {
xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> - %tu\n");
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops -- probebly a loop or
something");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy for accounting
purposes
record_route();
#Accounting if a call is started or ended
if (method=="INVITE" || method=="BYE" || method=="CANCEL") {
acc_rad_request("200");
}
# grant Route routing if route headers present
# loose-route processing. Decreases security slightly regarding
# forged messages, but decreases load.
if (loose_route()) {
xlog("L_NOTICE", "Loose Routing\n");
t_relay();
break;
};
======================
My understand is this should cause the ACK messages that are part of the
transaction to get routed. They should not pass beyond this block of code
right?
However, the SIP conversation goes as follows.
Gateway invites proxy
Proxy says 100 trying
Proxy says 100 trying
Proxy invites user
User says 100 trying
User says 180 Ringing
User says 180 Ringing
Thus far all is well
When the user picks up the phone
User tells proxy 200 ok
Proxy tells gateway 200 ok
Gateway tells proxy ACK
At this point the proxy should be relaying the ACK to the user, but it does
not. Instead due to timeout the user sends another 200 ok and that last
block of conversation repeats.
Any ideas why this would not be working?
(I tried to send this with a full SIP dialog but it got rejected as being
too big so this email is the chopped down version)
Hi all,
I was trying to configure my script for hosting multiple domains. I went through the admin's guide and understood the procedure of creating new subscriber table for each domain and serve them. But I would be very thankful if someone let me know how does automated multi-domain support works and how do we maintain multiple domains using automated support.
I think using domain table will help, but any information regarding serving multiple domains in an efficient manner will be very helpful.
Thanx a lot.
jayesh..
---------------------------------
Start your day with Yahoo! - make it your home page
Hi everybody,
I just finished adding support for BRANCH routes. The idea and frame
were partial ported from SER (thanks to Juha for extracting and adapting
patches). For those who are not familiar with this concept: branch route
is a route to be execute separately for each branch before being sent
out - changes in that route should reflect only on that branch.
I say partial because the features, implementation were improved.
Shortly here is a list:
- remove from grammar default BRANCH ROUTE definition - since this
is a 100% TM route, make no sense
- RURI may be changed from a branch route (the routing will
accordingly change); This capability is essential, since there are core
function which alter the RURI, functions which cannot be blocked in
specific route types - not handling this, may easyly lead to memory
faults or leaks.
- flags per branch - if flags are changed in branch_route, the
change will not be visible for other branches, but it will be visible
for all replies of that branch (it might be useful for per branch NAT
traversal)
- full AVP support in branch route
- more efficient handling of lumps (changes in request) - instead of
cloning and removing the entire lump tree for each branch, using lump
flags, after each branch, the original lump tree will be just cleaned
only what was added during branch route will be removed).
still pending: how to handle append_branch() core function.....I will
prefer to handle properly all core functions, since their execution
cannot be banned from specific routes.
For the moment only textops, avpops and xlog modules were enabled for
branch route. Other will come in the future.
regards,
Bogdan
PS: testing done only superficial (is quite late now).....any
help/reports are welcomed :D
Hello,
I'm still trying to learn more about SER. I've been using Asterisk to
manage virtual PBX services for different companies by using multiple
contexts within Asterisk. However, since I only use Asterisk with SIP
UAs and to communicate with ITSPs, I don't have the need to have all
the fancy features Asterisk offers, plus I have the additional
advantage of having the built-in NAT support in SER.
The question I have is if someone can point me to the right place
where I can see some sample configs that do more or less the things I
need or if someone would be willing to share some of those configs on
the list.
Basically, I need to have the ability to manage any number of virtual
PBX services where each virtual entity can manage their own extension
numbering, DIDs, outbound CLID. I would probably just continue using
Asterisk for voicemail services since I get the feeling that SEMS is
still "unstable" to go into production, even for voicemail services
only (unless told otherwise).
The idea is that even if two companies define two extensions 101 and
two extensions 102, when each company calls ext 101 or 102
internally, the call will be maintained within that company and would
not cross over to the other company. Also, having features like one
main number (DID) that could ring in multiple extensions
simultaneously (forking) and if no one answers, leave a message in
the general mailbox, or that each extension could have, additionally,
a DID so that they can be reached directly from the outside.
Any help would be greatly appreciated.
Thanks,
Waldo
