Hi there:
I want to test the SIP based endpoint. I'm wondering if there is any
open-source XCAP server available for testing, or if SER has the XCAP server
module?
Thanks so much!
henry
Hello,
I am trying to setup a voicemail solution based on the tutorial published on
voip-info.org called "Realtime Integration of Asterisk with OpenSER" (
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Op…)
and I face an issue. I followed the steps carefully, and I am about to start
running the openser server.
But when I try to launch the server, two lines are troublesome. My first
error is the following :
0(0) ERROR:avpops:fixup_write_avp: bad attribute name in param 2 <i:10>
0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 133
It is related to the following line of the openser.cfg :
avp_write("$ruri", "i:10");
Then, I comment the line, and another error appears :
0(0) ERROR:avops:fixup_pushto_avp: bad param 1; expected : $pseudo-variable
...
0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 200
It is realted to the line :
avp_pushto("$ruri", "i:10");
I have the dialplan writen in the tutorial and I run the latest stable
version of OpenSER : openser-1.1.0-notls
Hopefully someone encontered the same issue, and managed to solve it.
Thank you in advance,
Antoine
Hi !!!
I'm using OpenSER 1.0.1 and realized some strange behaviour
when SIP UA sends ReInvite Messages.
Initial INVITE shown below (this is OK)
UA A OpenSER UA B
_____________________________________________________________
--------INVITE------------> -----------INVITE------------>
<--------200 OK----------- <---------200 OK----------------
---------ACK-------------> -------------ACK--------------->
But when an UA sends a ReINVITE message I realized following behaviour.
This happens approximatly every 3rd or 4th ReInvite.
UA A OpenSER UA B
_____________________________________________________________
--------INVITE------------>
-----------INVITE------------>
-----------INVITE------------>
<--------200 OK----------- <---------200 OK----------------
<--------200 OK----------- <---------200 OK----------------
.
.
.
.
Is there any reason for OpenSER sending ReINVTE twice?
Any chance to avoid this?
Sincerly
Stefan
In an outbound call I am able to use
consume_credentials() to get rid of all
Authorization: Digest...
lines in the header, except for the ACK.
Can I strip the header from the ACK?
I can't seem to catch it.
-g
--
Greg Fausak
greg(a)thursday.com
Hi all,
I'm using Openser with siptrace and I'm getting the
following sip messages: Invite, Ack, Bye and Cancel.
The Openser is recording these messages into sip_trace
table (DB openser).
I would like to get "200 OK" sip message too. Is it
possible??
PS: My openser is working together freeradius and
mysql.
Thanks
Alberto
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I have the following config:
if (client_nat_test("3")) {
#IP address in contact is private IP
#Via is different to address client has contacted us from
#If so then setflag7 to indicate that sender is behind NAT
xlog("client nat test 3 returned true for r-uri <$ru> in
route 3 so will setflag7");
setflag(7);
xlog("contact before update <$ct>");
xlog("via before update <$hdr(via)>");
force_rport();
#Rewrite Contact: header to contain Sip client's public
IP:port
fix_nated_contact();
xlog("contact is now <$ct>");
xlog("via is now <$hdr(via)>");
};
However the value returned by $ct is the same before and after
fix_nated_contact(). Is that because I'm doing something wrong, or because
I've misunderstood fix_nated_contact or because the $ct variable doesn't get
updated by fix_nated_contact? If the latter, how can I find out the result
of fix_nated_contact? The same applies to force_rport and the via details.
Any advice appreciated.
Regards
Cameron
Hi
I want to overwrite the contact header field using textops but some UAs send only an asterisk as a contact (like below). Is this defined in protocol (I didn't find) ? I don't decide what to do . Do I reply with an error or use another field (from,to) as contact
thanks
REGISTER sip:sip.anywhere.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.49;branch=z9hG4bK8eaa0451724f9bf4
From: "test" <sip:11223344@sip.anywhere.com;user=phone>;tag=9969d7f94927b840
To: <sip:11223344@sip.anywhere.com;user=phone>
Contact: *
Call-ID: 0c55cf4a80b02370(a)10.0.0.9
CSeq: 100 REGISTER
Expires: 0
User-Agent: Grandstream HT286 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
Hi, All,
I am using SER 0.9.6, rtpproxy 0.3 and MySQL 4.1.11. All are running under
Fefora Core 4 on one machine.
Now, there is a very serious problem that I fork 24 children and each ser's
CPU usage is around 15%. Why? I check the count of registered users is only
about 600. And the concurrent call is about 20. Why the CPU usage is so
large? The idle of CPU is almost 0%.
I find the archive that there is a same problem on
http://lists.iptel.org/pipermail/serusers/2005-October/024575.html, but he
used PGSQL database. Not MySQL DB.
Could you help me? Thanks!
Regards,
Paco
Hi,
How may I append some digits (like "_digits") to the Request-URI of
INVITE messages?
With rewriteuser("_digits") an INVITE line like:
INVITE sip:user@domain:5060;transport=udp SIP/2.0
results in:
INVITE sip:_digits@domain:5060;transport=udp SIP/2.0
and I what to result this:
INVITE sip:user_digits@domain:5060;transport=udp SIP/2.0
Any suggestion?
Thanks,
Ricardo.