Hi all,
Can someone please help me in formulating a regular expression. I have an expression like this:
if(uri=~"^sip:1[0-9]*@)
This matches numbers like 15552221212, but also matches number like 15552221212#.
when i make it as:
if(uri=~"^sip:1[0-9]@*)
Again this matches any special characters after the number.
I just wanted a regular expression that will match only numbers and no alphabets or special characters.
Can someone please help me to formulate a regular expression for this.
Thanks in advance.
w/regards,
Jayesh
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Hello list.
I'm wondering if anyone has implemented a routing rule depending on the time of the day.
Is there a way to acomplish this?, maybe using AVP ? or something like that?
Hope that someone could help me..
Regards,
Ricardo Martinez.-
Is the field "Content-length" a required field in a NOTIFY message? I
see that it is part of almost every message from OpenSER except the
NOTIFY message. This is causing some issues with Gaim SIMPLe interface
which processes the header based on the header string "Content-Length".
Note Gaim does not care about the value associated with the header
string just the fact that the string "Content-Length" exists.
Therefore, since OpenSER isnt sending the string, Gaim is core dumping.
FYI on the core dump. Yes, Gaim still has a bug on how it is processing
what it sees as unknown SIP messages which actually is causing the core
dump, the real problem is the fact that Gaim is processing the header,
expecting to see the field "Content-length". So before I report this to
Gaim as a bug, I wanted to get insight into whether the field is a
required field.
Thanks......
Hello,
I got problem with the radius accounting, I am getting accounting packets
but they doesn’t mention start or stop time of call rather all the time
saying Acct-Status-Type = Failed
Here is one packet
Tue Nov 21 17:34:58 2006
Acct-Status-Type = Failed
Service-Type = IAPP-AP-Check
Sip-Response-Code = 487
Sip-Method = 1
User-Name = "211069021(a)xxx.pt"
Calling-Station-Id = "sip:211069021@xxx.pt"
Called-Station-Id = "sip:0969830612@xx.pt"
Sip-Translated-Request-URI = "sip:09643243612@22.18.31.10"
Acct-Session-Id =
"597816CF-EFCF-4AD8-B61D-F34301FE0B1B(a)192.168.0.47"
Sip-To-Tag = "38CA5520-DF9"
Sip-From-Tag = "26232679627249"
Sip-Cseq = "2"
NAS-Port = 5060
Acct-Delay-Time = 0
NAS-IP-Address = 127.0.0.1
Client-IP-Address = 127.0.0.1
Acct-Unique-Session-Id = "d26959cb062c7786"
Timestamp = 1164130498
I looked into the dictionary.ser which is present in
/usr/local/etc/ser/dictionary.ser and that dictionary had commented out
start and stop attributes. If I uncomment them and comment Failed attribute
then ser gets crashed.
According to ser-radius how to… .. there should be a dictionary.ser file in
/usr/local/etc/radiusclient-ng/ directory, but I tried different versions of
radiusclient-ng.. it doesn’t come with any of the version, though documents
mention it comes with radiusclient-ng library.
Though I tried many options with the dictionary.ser file by including it in
radiusclient-ng and in freeradius server, but no success.
I followed exactly the ser-radius how to but no luck . infact tried with
many other probable configure options but it is just not getting solved.
I am using ser-0.9.6 version
Please help
I will be very thank full
Lokesh
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.409 / Virus Database: 268.14.11/543 - Release Date: 11/20/2006
Hi,
I am working on a project which involves testing SER with some dummy clients.
I have downloaded and installed SER from iptel.org. I am using the very basic Hello.cfg configuration file available online.
I find that when SER gets a INVITE request from the caller, it does not deliver it properly to the client (callee). Some debugging revealed that it relays the request to port 5060 at the client rather than the one which was registered by the callee earlier. I tried to look at the source code of SER and found that the callee's contact information never seem to be looked up at any stage.
Is this due to some missing information in my configuration file?
Any feedback would be appreciated.
Thank you
-Kaushik
PS: I have attached (inline) the configuration file
ser.cfg
------------------------------------
debug=9
fork=yes
log_stderror=yes
listen=128.42.3.3
port=5060
children=4
check_via=no # added
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
loadmodule "/scratch/ser-0.9.7-pre1/modules/sl/sl.so" # Stateless replies
loadmodule "/scratch/ser-0.9.7-pre1/modules/tm/tm.so" # Transaction management
loadmodule "/scratch/ser-0.9.7-pre1/modules/rr/rr.so" # Routing and record-routing
loadmodule "/scratch/ser-0.9.7-pre1/modules/maxfwd/maxfwd.so" # Max fowards check
loadmodule "/scratch/ser-0.9.7-pre1/modules/usrloc/usrloc.so" # User location support
loadmodule "/scratch/ser-0.9.7-pre1/modules/registrar/registrar.so" # Registrar (requires userloc and sl)
loadmodule "/scratch/ser-0.9.7-pre1/modules/textops/textops.so" # Message textual operations
modparam("usrloc", "db_mode", 0) # 0 forces registration data to be stored in memory
modparam("rr", "enable_full_lr", 1)
route {
# ------------------------------------------------------------------------
# Sanity Check Section
# ------------------------------------------------------------------------
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
break;
};
if (msg:len > max_len) {
sl_send_reply("513", "Message Overflow");
break;
};
# ------------------------------------------------------------------------
# Record Route Section
# ------------------------------------------------------------------------
if (method!="REGISTER") {
record_route();
};
# ------------------------------------------------------------------------
# Loose Route Section
# ------------------------------------------------------------------------
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
# ------------------------------------------------------------------------
# Call Type Processing Section
# ------------------------------------------------------------------------
if (uri!=myself) {
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
if (method=="ACK") {
route(1);
break;
} else if (method=="REGISTER") {
route(2);
break;
};
lookup("aliases");
if (uri!=myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
append_hf("P-hint: userloc applied\r\n");
route(1);
}
route[1] {
# ------------------------------------------------------------------------
# Default Message Handler
# ------------------------------------------------------------------------
if (!t_relay()) {
log(2, "Failed t_relay");
sl_reply_error();
};
}
route[2] {
# ------------------------------------------------------------------------
# REGISTER Message Handler
# ------------------------------------------------------------------------
if (!save("location")) {
sl_reply_error();
};
}
Hi,
Thanks for your reply but it will not match only two digits as I have specified the range( [0-9] ). Actually my primary requirement is it should not match any alphabets or special characters with the numbers.
I want that only the number should be matched.
w/regards,
Jayesh
----- Original Message ----
From: Alexandr Dubovikov <shurik(a)start4.info>
To: Jayesh Nambiar <voip_freak(a)yahoo.co.in>
Cc: users(a)openser.org
Sent: Wednesday, 29 November, 2006 1:15:51 PM
Subject: Re: [Users] regular expression
On Wed, Nov 29, 2006 at 12:58:44PM +0530, Jayesh Nambiar wrote:
> Hi all,
> Can someone please help me in formulating a regular expression. I have an expression like this:
> if(uri=~"^sip:1[0-9]*@)
> This matches numbers like 15552221212, but also matches number like 15552221212#.
>
> when i make it as:
> if(uri=~"^sip:1[0-9]@*)
not really, in this regexp your number can have only 2 digits and first one
should ne "1".
> Again this matches any special characters after the number.
>
> I just wanted a regular expression that will match only numbers and no alphabets or special characters.
> Can someone please help me to formulate a regular expression for this.
if(uri=~"^sip:1[0-9]+@*)
>
> Thanks in advance.
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
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Yahoo! India Answers: Share what you know. Learn something new
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Hey All,
I am looking for someone who can give me some assistance with a
building an openser configuration that is able to handle the following:
1) Multiple domains.
2) Mediaproxy for all clients (not just clients behind NAT).
3) RADIUS accounting for calls.
4) Least Cost Routing for selecting the cheapest upstream link.
5) Trusted (non-authenticated) upstream providers for receiving calls.
6) Call Redirection (all calls, busy, no answer) with redirected calls
billed to the redirecting party.
7) Some network features, like voicemail, talking clock, etc, sent to an
Asterisk machine.
I currently have a configuration that does (1) through (5), but have had
no luck in even getting call redirection to work, let alone getting the
billing right.
I'm having some issues, like:
- Getting two replies to REGISTER messages which causes Asterisk clients
to print a message "Got authentication request (401) on unknown
REGISTER to <...>"
- Sometimes I get a storm of CANCEL or BYE messages when a call is
ending. I have absolutely no idea where the messages are coming from,
whether its the client or upstream, and what message I'm sending
that's causing it.
The configuration I have made is just from reading the docs, looking at
the howtos and tutorials and from guessing what should be done. I think
the problems I'm having are related to not really understanding what
should be happening 100% and having a config that is a lot more
complicated than it needs to be.
Anyone willing to give me some assistance? I can send a password
sanitised copy of my current config directly if it would help.
Regards
Darryl
i use the strip() function to strip the prefix when call out
for example, prefix 0 to call out
there are my sip invite message
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
INVITE sip:0928117466@202.111.222.76:5060 SIP/2.0
Record-Route: <sip:211.111.222.102;r2=on;lr=on;ftag=694064c4>
Record-Route: <sip:211.111.222.102;transport=tcp;r2=on;lr=on;ftag=694064c4>
Content-Length: 324
Content-Type: application/sdp
Via: SIP/2.0/UDP 211.23.176.102;branch=z9hG4bK4eec.ecf680e4.0;i=1
Via: SIP/2.0/TCP 192.168.123.5:5060;received=220.132.138.7
;branch=z9hG4bK69486617
To: <sip:00928117466@211.111.222.102>
From: "joepass" <sip:joepass@211.111.222.102>;tag=694064c4
Supported: timer
Call-ID: 95746504-39004973-1533c430-d8319d29(a)192.168.123.5
CSeq: 26589 INVITE
User-Agent: IP SIP Phone/2.1.3
Max-Forwards: 69
Session-Expires: 1800
Allow: UPDATE,INFO,MESSAGE,REFER,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL
Authorization: Digest nonce="456ba868fbd62c72ca16fcdd04678168a8fa0683",
username="joepass", realm="votel-tech.com", uri="
sip:00928117466@211.111.222.102",
response="73a8c869c2a42a12f0d920c2a7d6f068"
P-IPRAuth: votel-tech.com
Contact: <sip:joepass@220.111.222.7:1070>
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
in the example, the real number is 0928117466
add prefix 0, so the final number is 00928117466
I see the INVITE part the id has strip the prefix 0
but
To: <sip:00928117466@211.111.222.102>
and the Authorization part(uri) still keep on 00928117466
My gateway seems to use this information to call out.
So it cause some error respond.
If i call to the gateway directly(not through openser with number
0928117466), it works.
How can i strip the "To" and "Authorization" part uri.
thanks...
Sam
Hi,
We can use the dispatcher module, but it will only load balance the SIP traffic. Correct?
I tried to use this http://mediaproxy.ag-projects.com/ with one ser server in front of two Asterisks. But I guess that this tool will only work with multiple ser servers...
Is there a solution to make RTP/SIP load balancing with asterisks ?
Thanks for your help,
Thomas
-----Message d'origine-----
De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Daniel-Constantin Mierla
Envoyé : vendredi, 13. octobre 2006 11:25
À : yusuf
Cc : users(a)openser.org
Objet : Re: [Users] OpenSER in front of Asterisk
Hello,
if you want to do load balncing, have a look at dispatcher module, it is
intended to do that. Does not mather if behind are asterisks or other boxes.
Cheers,
Daniel
On 10/10/06 10:49, yusuf wrote:
> Hi all,
>
> I just joined this mailing list today :)
> I'm sure you guys have heard this question before, but what I want to
> do is put OpenSER in front of Asterisk to achieve load balancing. I
> realise this entails more than it means, and will probably open up a
> world of pain, I just want to know what others experiences of this
> are, and any comments.
> Also, with 1.1.0 being released, and with this new module called
> 'path', will this be of any help
>
>
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