[OpenSIPS-Users] One Way Audio
Hi Irina,
Thanks for reply. After looking in forums I observed that on opensips
version 1.6 has a bug and its bug fix is uploaded on svn. I recompile svn
version 1.6 and test it and working ok.
But now I'm facing weird problem, while using non-svn version 1.6 I was able
to call to my asterisk boxes and media was passing on both ways. But when I
recompile svn version 1.6 and make a call there is only one way voice from
eyebeam to twinkle i.e.
eyebeam - opensips -- asterisk -- twinkle
twinkle can hear from eyebeam side
---
eyebeam can't hear from twinkle side
Opensips and Asterisk both hosted on Public IPs and UAC are located at
private network. Firewall is permitted on both servers and I'm using stun
for my UAC.
Kindly advise to sort this problem? But I don't understand why was media is
passing both ways when using non-svn version?
Further added, I am also using module dispatcher.
Date: Wed, 06 Jan 2010 16:36:44 +0200
From: Irina Stanescu istane...@opensips.org
Subject: Re: [OpenSIPS-Users] How to increase cache in opensips tables
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 4b449ffc.5050...@opensips.org
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Ahmed,
Firstly, I need to see the log so I could understand better the error
you get. I don't think the problem is that the cache is too small.
Also, you cannot use 0 for the group id, the documentation says:
group_id
This argument represents the group id to be matched. It can be an
integer string or a string pvar. If the group_id argument is 0, the
query can match any group in the cached address table
Secondly, as the name suggests, the ip column is reserved for IPs only.
You cannot add domain name addresses to this column.
Regards,
Irina Stanescu
Ahmed Munir wrote:
Hi,
I'm using permission module's function check_source_address(), the
problem I'm facing is that I can not add not than more 8 IPs in
address table, but I want to permit more than 100 IPs. I only want to
use these IPs on group 0 what I am using. When I enter more than 8 IPs
in address table and make a call, I observe a message i.e. not found
in hash table.My opensips.cfg configuration for check_source_address()
is listed below;
if (is_method(INVITE) check_source_address(0)) {
log(INVITE###);
ds_select_domain(1,4);
forward();
route(1);
log(#END);
setflag(1);
}
Kindly advise me how to increase cache of OpenSIPs database tables so
I can reslove my case. Further added, how can I enter domain name in
'ip' column section of address table i.e. abc.com http://abc.com
can't be used and gives me an error, kindly advise this well.
--
Regards,
Ahmed Munir
--
Regards,
Ahmed Munir
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Re: [OpenSIPS-Users] OpenSIPS 1.6 drouting: drouting:dr_load_routing_info: route 1 does not exist
Hi Mike, the routeid field points to a script route to be executed when the rule does match - see http://www.opensips.org/html/docs/db/db-schema-1.7.x.html#GEN-DB-DR-RULES. so, it must match the name of a defined script route. if you do not need this feature, let the field in DB NULL. Regards, Bogdan Mike O'Connor wrote: Hi All I'm trying to get drouting working, the issue is I can not work out which tables and field its complaining about when it give the error. 'drouting:dr_load_routing_info: route 1 does not exist' In 'dr_rules' I have a routeid of 1 for a number of rows, and if I change this to a 2 or 5 then the above error changes. So the issue must be that I have not place a route some where with the id of 1, problem I can not find where to add this record. Any help greatly appreciated. Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] need help configuring TLS server
hi all, I am looking for someone how can help me configure opensips with TLS enabled, please contact me directly, thanks, nir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need help configuring TLS server
Hi Nir, Check first http://www.opensips.org/html/docs/tutorials/tls-1.4.x.html Regards, Bogdan nir elkayam wrote: hi all, I am looking for someone how can help me configure opensips with TLS enabled, please contact me directly, thanks, nir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dialog Problem
Hi all, I have installed opensips 1.6 . When i make a call between 2 users (3 concurrent calls), when the calls is disconnect only the dialog of the latest call is deleted from the dialog table. other 2 calls dialog hang around in the DB. Another strange behavior seen is that the callid of BYE's of the first 2 calls are completely different from the invite they used to initiate the call -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need advice on B2b
HI Anca
it is clear now :)
i am trying to learn the opensips,
Thank you
Ha`
--- On Wed, 1/6/10, Anca Vamanu a...@opensips.org wrote:
From: Anca Vamanu a...@opensips.org
Subject: Re: [OpenSIPS-Users] need advice on B2b
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Wednesday, January 6, 2010, 3:14 AM
Hi Ha`,
The excerpt from your script shows that you don't have a good
understanding of the opensips scripting logic. First, the route
block will only be called for SIP Requests.
Calling this is not right:
if(status==200)
route(b2b_reply);
The replies will go into reply route blocks. You can have a default
reply route ( one without an index or with index 0), or you can specify
a certain reply route for a request by calling t_on_reply.
Second, you don't understand what happens with b2b request and replies.
It is explained in the documentation:
The requests and replies that are received by the B2BUA server,
belonging to the dialogs it is handling will not go into the script as
normal request do. The reason for this is that this are not normal
requests where the server is a proxy, but the server is an endpoint in
the dialog and therefore they should not go through the same routes.
However, it is normal for this request to be seen from the script and
allow the script writer to do the processing it desires based on them.
For this, it is possible to define two special B2B routes - one for
requests and one for replies. The routes are of type route
and have their name defined in the modules parameters script_req_route
and script_reply_route.
In other words, there are two important things:
1. the B2B requests/replies will not go into the default request/reply
route block.
2. the b2b_request/b2_reply route will be called automatically for
every request/reply targeted to the b2b agent
So, for your script, you don't need this lines:
if(is_method(INVITE)) {
route(b2b_request);
t_on_reply(2);
}
else
if(status==200)
route(b2b_reply);
Hope this made things a bit clearer.
Regards,
--
Anca Vamanu
www.voice-system.ro
ha do wrote:
HI Anca
i am trying to use the b2b_request + b2b_reply
route{
...
if(is_method(INVITE) !(src_ip == 192.168.1.249
src_port ==5060))
{
if (! t_newtran()){
sl_reply_error();
exit;
};
b2b_init_request(top hiding);
exit;
};
route(1);
}
route[1] {
if(is_method(INVITE)) {
route(b2b_request);
t_on_reply(2);
}
else
if(status==200)
route(b2b_reply);
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[b2b_request] {
xlog(b2b_request cucku ($ci)\n);
force_rtp_proxy();
}
route[b2b_reply] {
xlog(b2b_reply cucku ($ci)\n);
force_rtp_proxy();
}
i get the errors :
ERROR:nathelper:force_rtp_proxy: Unable to parse body
and
DBG:tm:utimer_routine: timer routine:4,tl=0xb615e9a8 next=(nil),
timeout=2900
DBG:tm:timer_routine: timer routine:3,tl=0xb615c6f4 next=(nil),
timeout=29
DBG:tm:delete_handler: removing 0xb615c690
DBG:tm:delete_cell: delete_cell 0xb615c690: can't delete -- still
reffed (1)
===
DBG:core:get_hdr_field: cseq CSeq: 2 INVITE
DBG:core:parse_headers: flags=8
DBG:tm:t_reply_matching: hash 21530 label 76806763 branch 0
DBG:tm:t_reply_matching: REF_UNSAFE: after is 2
DBG:tm:t_reply_matching: reply matched (T=0xb615e85c)!
DBG:tm:t_check: end=0xb615e85c
DBG:tm:reply_received: org. status uas=0, uac[0]=100 local=2
is_invite=1)
DBG:tm:t_should_relay_response: T_code=0, new_code=180
DBG:tm:local_reply: branch=0, save=0, winner=0
DBG:tm:local_reply: Passing provisional reply 180 to FIFO application
DBG:tm:run_trans_callbacks: trans=0xb615e85c, callback type 1024, id 0
entered
DBG:b2b_entities:b2b_parse_key: hash_index = [111] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:core:get_hdr_field: content_length=0
DBG:core:get_hdr_field: found end of header
DBG:b2b_entities:b2b_tm_cback: Received a reply with statuscode = 180
DBG:core:parse_headers: flags=
DBG:b2b_entities:b2b_new_dlg: 'To' header ALREADY PARSED:
sip:1...@192.168.1.249
DBG:b2b_entities:b2b_new_dlg: Not an initial request
DBG:core:parse_to_param: tag=bfad35cdb22f09f741816636d344f54b-19f0
DBG:core:parse_to: end of header reached, state=29
DBG:core:parse_to: display={},
ruri={sip:0873000...@192.168.1.249;user=phone}
DBG:core:print_rr_body: current rr is sip:192.168.1.249;lr=on
DBG:core:print_rr_body: out rr [sip:192.168.1.249;lr=on]
Re: [OpenSIPS-Users] Need help on flag in usrloc
Hi Bogdan
it is clear now :)
Thank you
Ha`
--- On Wed, 1/6/10, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Wednesday, January 6, 2010, 3:51 AM
Hi Ha,
the two flags are different and may have different values - one is used
as NAT marker, the other one is used as SIP-based pinging marker.
so, you can use different flags and both of them will be saved in cflag
mask.
Regards,
Bogdan
ha do wrote:
Hi Bogdan
got it :)
1 more question about the flag
modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 7)
the modem ADSL will close the port after 3 mins(some minutes), so
Opensips should send OPTION message(sip ping) to modem to keep port
that should open for UA
the sipping_blag of nathelper module should be the same value as
nat_bflag of usrloc ? or the cflag of usrloc just has a value??
Thank you
Ha`
--- On *Tue, 1/5/10, Bogdan-Andrei Iancu /bog...@voice-system.ro/*
wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, January 5, 2010, 7:20 AM
Hi Ha,
the NAT branch flag you use is 7 (nat_bflag) and in usrloc you
find the
branch flags in the cflags (contact flags) field. The cflags is a
mask
with all the branch flags: 192 = 128 (2^7) + 64 (2^6)
Regards,
Bogdan
ha do wrote:
Hi all
i am successfull to check the UA behind NAT but i dont know what
value
of the flag will be stored in the usrloc
Could someone please let me know the value of Nated UA flag,
that is
stored in usrloc
my config :
modparam(nathelper, natping_interval,180)
modparam(nathelper, ping_nated_only, 1)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from, sip:cu...@kamailio.org
/mc/compose?to=cu...@kamailio.org)
modparam(registrar|nathelper, received_avp, $avp(i:80))
modparam(usrloc, nat_bflag, 7)
route{
route(4);
if (method==REGISTER)
{
if (isflagset(5)) {
setbflag(6);
setbflag(7);
}
if (!save(location))
sl_reply_error();
exit;
}
}
route[4]{
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
mysql select * from location\G
*** 1. row ***
id: 12
username: 1000
domain: NULL
contact: sip:1...@192.168.1.2:12280;rinstance=a4752f45bdc3ddd3
received: sip:210.245.35.150:12280
path: NULL
expires: 2010-01-05 17:48:56
q: -1.00
callid: YjE1NWJmYWYyYWMzZDg2ZDc5MjY0NDQyMGE5NDEwM2E.
cseq: 2
last_modified: 2010-01-05 16:48:56
flags: 0
cflags: 192
user_agent: eyeBeam release 1004p stamp 31962
socket: udp:118.69.193.198:5060
methods: 5951
Thank you
Ha`
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[OpenSIPS-Users] need help on mediaproxy that cannot start
Hi all i follow the instruction : http://www.smartvox.co.uk/serfaq_install_mediaproxy2.htm my centos version: [r...@centos-cucku application]# uname -a Linux CentOS-Cucku 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux i get the error when start the mediaproxy on Centos : [r...@centos-cucku application]# /usr/bin/media-dispatcher Traceback (most recent call last): File /usr/bin/media-dispatcher, line 12, in ? from application.process import process, ProcessError File /usr/lib/python2.4/site-packages/application/process.py, line 12, in ? from application import log File /usr/lib/python2.4/site-packages/application/log/__init__.py, line 12, in ? from application.log.extensions import twisted File /usr/lib/python2.4/site-packages/application/log/extensions/twisted/__init__.py, line 4 from __future__ import absolute_import SyntaxError: future feature absolute_import is not defined please help thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] INVITE with unknown udp port number
Dear all, I am facing the following problem during INVITE transaction. Since my opensips-1.5.0 has forwarded INVITE with unknown udp port number to X-Lite as SIP UA, Port unreachable occurs at SIP UA and INVITE transaction fails. Could anyone teach me why the unknown udp port number is set and how this problem should be fixed? Regards, Yagishita ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
