Hello,
I'd like to permit our subscribers to have several aliases for outgoing
calls (basically, several number in the PSTN). I think they could set this
in the header From of the INVITE to us and we would identify them,
authenticate them and finally relay that same alias (and not the subscriber
name) to the GW (currently we permit them to have a single alias for
outgoing using a field in the subscriber table).
I can see the module alias_db can be used to manage aliases, but I
understand it can only be used for incoming calls as it provides just a
single function alias_db_lookup that operates on the Request-URI.
Any ideas?
regards,
takeshi
Hi everyone,
I have a doubt about the function avp_pushto(). When I do the following:
...
avp_pushto("$ru", "$avp(s:devices)/g");
...
If $ru is an array with more than 1 position, avp_pushto works fine, but if
both $ru and $avp(s:devices) have only 1 position, the function avp_pushto
replace $ru with the new value. It cause loosing of the previous $ru.
I believe that avp_pushto function shouldn't replace a value. Any idea?
I'm using OpenSER 1.3.2
Thanks in advance.
Pablo Saro
psaro at google dot com
I will apologize in advance if this is the wrong forum or if this is an
embarrassing newbie question.
I am currently using an Trixbox/Asterisk system to connect a Rolm PBX to
a ShoreTel phone system via SIP. The asterisk box doesn't correctly
support the REFER method, so I'm hoping to use SER to proxy my SIP
connections to allow the Rolm users to function correctly with transfers
and conferencing.
I have installed and configured my SER system using the getting started
documentation. I have successfully interconnected my SER system with my
Asterisk and with a Cisco 2801 and can place and route calls as I would
expect. When I try to place a call through SER to my ShoreTel system, I
get the standard "SIP/2.0 404 Not Found". If I compare captures of my
successful SIP connection from my Asterisk environment to a capture of
the SER attempt, the only real difference I see is that the INVITE TO:
field has the IP of my SER system instead of the ShoreTel system. The
actual INVITE uri is identifying the ShoreTel device, but the TO field
has the SER address. The Asterisk connection has the address of the
ShoreTel device in both places.
I have tried to look through the docs and searched online, but I did not
find any resources on this question. I would appreciate any feedback
that could point me in the right direction.
Here is a ser -V from my SER system
version: ser 0.9.7-pre1 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $
main.c compiled on 14:05:21 Jul 16 2008 with gcc 4.1.2
I route my calls to the ShoreTel system through the basic PSTN route
from the getting started sample:
route[5] {
#
-----------------------------------------------------------------
# PSTN Handler
#
-----------------------------------------------------------------
rewritehost("my.IP.addr.here"); # INSERT YOUR PSTN GATEWAY IP
ADDRESS
avp_write("i:45", "inv_timeout");
t_on_failure("1");
route(1);
}
Thank you for any feedback in advance.
Hello
I have installed openser 1.2.3 and mediaproxy 1.9.0 on my server.
Im having problems with mediaproxy dispatcher time response. Is taking too
much time to answer.
I have the following message in the logs
proxydispatcher[9601]: execution time: 2183.73 ms
Could you please help me on this issue?
Thanks in advanced
Ariadne L. Ramos Solís
Depto. Ingeniería
Aprovisionamiento y Señalización VoIP
Galaxy Communications Corp.
Tel. 2000117
e-mail aramos(a)clarocom.com
hi list ,
I am trying to connect openser with asterisk, but when I make calls to the pstn, it shows me a 407 Proxy Authentication Required , am trying to make groups of calls with openser that some extensions
have permission of making calls to the pstn, others that alone they
call internally and international. I have created 3 groups local, int,
international, the extensions that are of the local group have
permission I believe me, according to the theory to be able to make
local calls,for this I am using asterisk, but this he throws me the
message 407.
The extension 112 with the address ip 192.168.10.26 belongs to the
local group, and he makes an intent of calling to the pstn through
asterisk, but this he throws me the message 407 I don't know because
thank you for your help
+0.399494 192.168.10.26:5060 -> 192.168.10.1:5060
INVITE sip:2642006@192.168.10.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.10.26:5060;branch=z9hG4bKb7bfb742a0e47ab4
From: "Ventas" <sip:112@192.168.10.1;user=phone>;tag=ca46913c384c257c
To: <sip:2642006@192.168.10.1;user=phone>
Contact: <sip:112@192.168.10.26:5060;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: fb973f9069a09ad7(a)192.168.10.26
CSeq: 59632 INVITE
User-Agent: Grandstream GXP2000 1.1.5.15
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 382
v=0
o=112 8000 8000 IN IP4 192.168.10.26
s=SIP Call
c=IN IP4 192.168.10.26
t=0 0
m=audio 5004 RTP/AVP 0 18 3 2 99 9 4 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/16000
a=rtpmap:4 G723/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
#
U +0.000205 192.168.10.1:5060 -> 192.168.10.26:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.26:5060;branch=z9hG4bKb7bfb742a0e47ab4
From: "Ventas" <sip:112@192.168.10.1;user=phone>;tag=ca46913c384c257c
To: <sip:2642006@192.168.10.1;user=phone>;tag=d7e793d797690be5ce5420203095040f.2cd4
Call-ID: fb973f9069a09ad7(a)192.168.10.26
CSeq: 59632 INVITE
Proxy-Authenticate: Digest realm="192.168.10.1", nonce="48802341954eb0c26cd5eb2dd85f375325fb4a7d", qop="auth"
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0
I wonder that how openser parse the source port from the request. Does the parser function get the port from the header of packet(TCP/UDP) or from the content(by parsing Via header or something) of request message.
Regards,
Kevin
_________________________________________________________________
这里好多好玩的视频，用鼠标点到视频看看，有惊喜！
http://cnweb.search.live.com/video/results.aspx?q=%E5%A5%A5%E8%BF%90%E5%9C%…
But after forward, new invite will be generated so avp will be lost, isn't
it?
On Fri, Jul 18, 2008 at 7:00 PM, kokoska rokoska <kokoska.rokoska(a)post.cz>
wrote:
>
>
>
> Ruchir napsal(a):
>
>> This is for explicit call forwarding. I'm testing call forwarding from
>> linksys pap2 call forward features which generates "302 moved temporarily"
>> message.
>>
>>
> I too :-)
> May be I don't understand what you are looking for...
>
> From my point of view it is simple:
> 1. If IS present Diversion header in 302 reply, than push it to avp
> 2. If Diversion header IS NOT present, than use "unknow" reason - like in
> SS7...
>
> Best regards,
>
> kokoska.rokoska
>
>
> On Fri, Jul 18, 2008 at 6:20 PM, kokoska rokoska <kokoska.rokoska(a)post.cz<mailto:
>> kokoska.rokoska(a)post.cz>> wrote:
>>
>>
>>
>>
>> Ruchir napsal(a):
>> > I'm using uac_redirect module to handle redirect and accounting from
>> > openser. Redirect works fine so as cdr but I'm not getting how to
>> write
>> > redirect reason in cdr. I didn't find any way to find and store
>> redirect
>> > reason(call forward, busy, no answer) in CDR. Does anyone know
>> how to do it?
>> >
>> >
>>
>>
>> I'm using something like this (shortened):
>>
>> in config:
>>
>> modparam("acc",
>> "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)")
>>
>>
>> in routing script:
>>
>> $avp(s:acc_state) = "cfu";
>> avp_printf("$avp(i:901)", "$avp(s:caller_uuid)|$avp(s:acc_state)");
>>
>>
>> Works very well :-)
>>
>> Hope this helps, best regards,
>>
>> kokoska.rokoska
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)lists.openser.org <mailto:Users@lists.openser.org>
>> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
i'm want to implement watcher application in openser ..
but did find any idea and what fuctionality wathcer have.
can any one help me regarding how i got help regarding this.
~suresh
Hi!
How is the overall experience like re. deploying openser with mysql
clusters? Are there gotchas etc that need to be taken care of? (For
example, a 2006 article [1] says that "The MySQL NDB engine currently
runs its database completely in memory. This means that you have to be
able to fit your database in memory." But this is not documented as a
limitation in mysql faq.) Non clustered experiments were occasionally
catastrophic, so can't risk taking that route again.
Somebody suggested using drbd [2], but I did not quite like it -- it
needs kernel modules, has more administrative overhead, is tied to
linux, it is not clear how consistency (ie, state of in-memory data
and on-disk data) is taken care of, and in any case ndbcluster appears
to be the clustering solution with official stamp of approval.
(Related - is postgres any better? Apparently postgres' clustering
capabilities are "better", but then cdrtool etc seems to be written
with mysql as the primary target.)
Thanks,
Sajith.
[1]
http://www.oreillynet.com/pub/a/databases/2006/02/16/ha_mysql_cluster.html
[2] http://www.drbd.org/
--
"the lyf so short, the craft so long to lerne."
-- Chaucer.
Hi, I want to do a case insensitive match, for example for Privacy header that
can contain "id", "ID", "Id"... and other values.
I need something as:
if ($hdr(Privacy)=~"id"/i) {
but this obviously fails. Which is the way to do it?
Thanks.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
