Hi,
I have a problem.
I have a working openser server, but I want and a media server, so I decide to use sems.
I put everything in this way:
Openser <-------> Ser<--------->sems
All things are working on the same server.Openser works on 5060 port, and ser on 6060 port.
In openser.cfg I redirect to ser in this way:
if (uri=~"sip:100@.*") {
t_relay("localhost:6060");
exit;
};
and in ser.cfg:
if (uri=~"sip:100.*@") {
if(!t_write_unix("/tmp/am_sock","conference")) {
t_reply("500","error contacting sems");
};
break;
};
When I dial 100 the logs of the ser contain:
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SIP Request:
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: method: <INVITE>
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: uri: <sip:100@domain.com>
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: version: <SIP/2.0>
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=1
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: Found param type 232, <branch> = <z9hG4bK2cec.f018b0b5.0>; state=16
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=5
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: Via found, flags=1
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: this is the first via
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: After parse_msg...
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: preparing to run routing scripts...
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=128
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: Found param type 232, <branch> = <z9hG4bKOUUByO4Kp1q4sDNU>; state=16
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=5
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: Via found, flags=128
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: this is the second via
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG:maxfwd:is_maxfwd_present: value = 69
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DBG:maxfwd:process_maxfwd_header: value 69 decreased to 16
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=8
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_param: tag=6mVbY0KQFDRYupVP
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=29
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=256
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: end of header reached, state=9
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: get_hdr_field: <To> [29]; uri=[sip:100@domain.com]
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: to body ["100" <sip:100@voip.cts.md> ]
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: get_hdr_field: cseq <CSeq>: <1> <INVITE>
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: get_hdr_body : content_length=232
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: found end of header
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: find_first_route: No Route headers found
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: loose_route: There is no Route HF
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==9 && [domain.com] == [127.0.0.1]
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==11 && [domain.com] == [X.X.X.X]
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==10 && [domain.com] == [172.17.6.2]
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==9 && [domain.com] == [127.0.0.1]
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==11 && [domain.com] == [X.X.X.X]
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if host==us: 11==10 && [domain.com] == [172.17.6.2]
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: grep_sock_info - checking if port 6060 matches port 5060
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_self: host != me
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: t_newtran: msg id=3 , global msg id=2 , T onfiltered=0xffffffffffffffff
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=-1
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=60
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: t_lookup_request: start searching: hash=52930, isACK=0
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: RFC3261 transaction matching failed
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: t_lookup_request: no transaction found
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SER: new INVITE
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: parse_headers: flags=-1
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_via_address(172.17.6.2, 172.17.6.2, 0)
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: WARNING:vqm_resize: resize(0) called
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: reply sent out. buf=0x5d8838: SIP/2.0 1..., shmem=0x2b9b012ee768: SIP/2.0 1
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: _reply_light: finished
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: mk_proxy: doing DNS lookup...
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: get_record: lookup(_sip._udp.domain.com, 33) failed
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: sip_resolvehost: no SRV record found for voip.cts.md, trying 'normal' lookup...
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: check_via_address(172.17.6.2, 172.17.6.2, 0)
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_to_tail_of_timer[4]: 0x2b9b012ec368
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG: add_to_tail_of_timer[0]: 0x2b9b012ec388
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: SER: new transaction fwd'ed
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: DEBUG:destroy_avp_list: destroying list (nil)
Oct 4 12:26:59 voip /usr/local/sbin/ser[31088]: receive_msg: cleaning up
Please show me where is the problem, in openser or ser or sems?
Thanks.
Ion M.
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hi all
anyone have idea to use SIPP and how to you this tool in brief
thanks
vijay
---------------------------------
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Hello,
I have a question about a good way to handle retransmissions.
I see that t_check_trans() function is able to tell me if the request is a retransmission or not.
So my question is: should I apply the whole request-processing logic to retransmissions
or I can simply do something like the following:
if (is_method("ACK|CANCEL")) {
if (t_check_trans()) {
t_relay();
} else {
xlog(L_NOTICE,"$rm: No matching transaction. dropping");
exit;
}
}
if (t_check_trans()) {
xlog(L_NOTICE,"$rm: is a retransmission");
t_relay();
}
I.e. in the above code I assume that t_relay() would be able to match the request
to the transaction and absorb it like it does for ACKs and CANCELs without
applying whatever modification was done to original request.
Is it true?
Thanks,
Michael
Hello all,
I just updated the SST module to be fully compliant with the RFC. Here is a
list of the biggest changes:
1) The script function sstCheckMin() will now add the Min-SE: header and send
the 422 reply if the Session-Expire: header in the INVITE is too small. (with
the argument flag set to true) The old sstCheckMin() function would return
true stating that the Min-SE: value was too small, but the RFC states the 422
reply can *not* be sent without stating your Min-SE: value.
2) The Min-SE: and the Session-Expires: headers will be updated to your min_se
value if in the initial request if too small.
3) If the UAC does not request SST support, the SST module will request it on
behave of the PROXY.
4) The PRACK issue with the dialog module has been worked around. PRACK will
clobber the dialog expires lifetime value but the SST module will fix it.
5) If the UAS does not support SST, the SST module will request the UAC
refresh on behave of the PROXY.
6) reINTITE and UPDATE are now supported as refreshers to an active timed
session.
7) The introduction of the sst flag parameter. This is keeping with the idea
that the module will not do anything to a message unless asked for by the
script (openser.cfg) file. You now have to setflag(sst_flag) just like you
have to setflag(dialog_flag) for the module to act on a message.
8) Code clean up and commenting of all functions and beefed up the user
documentation.
Hope you find this module usefull,
ronw (karwin)
--
Ron Winacott - SOMA Networks, Inc.
---
Chaos, panic and disorder...my work here is done.
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How may I set some flag that isn't reset after some call leg has finished?
Every flag that I rise with setflag() gets always reset after that call
leg finishes!!
Thanks in advance,
Ricardo.
We just installed CDRtool over ser v.0.9.3 and appeared to install properly. However when we pull up the web portion, we get the initial login page. Login in with the default admin accounts and it takes us to the EULA. After agreeing we are taken to a mostly blank page with 2 frames, 1 pointing to callsearch.phtml and 1 pointing to blank.phtml. No errors are given in reference to database or access. Help?
Thanks in advance
Chris
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.407 / Virus Database: 268.13.1/466 - Release Date: 10/7/2006
Hi,
I recently experimented with TLS on a server that was initially enabled
without openser and it worked fine. Once I remade with TLS and restarted
SER, I noticed very poor reception on regular analog (w/ATA) and SIP phones,
such as fading out completely on audio, one-way audio issues again, or
fading in/out with regarding to sound or no sound at all and just dead,
lapsing air, so I concluded that OpenSER with TLS .cfg file is restricted to
TLS ONLY. Although I aware that TLS is to be strictly used with TLS enabled
software, I thought I would be able to use the same .cfg file and UAs that
use particular technologies would still work the same. How should it have
actually worked? Is this the norm?
Now, I have the TLS enabled Server on an entire different machine so that I
can test more thorougly but need to route it with the main OpenSER/Asterisk
system to determine how TLS functions using the PSTN. So I am now hoping to
experimentally have users wishing secure lines routing thru the one server
and then to another system, but sharing the same database as the first
server. And I am hoping that audio this way will not be an issue. Is this
the norm?
Thanks for any enlightenment.
Tracy
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i tried to checkout rel_1_1_0 to a new host and got:
cvs -z3 checkout -r rel_1_1_0 sip-server
...
cvs checkout: Updating sip-server/modules/imc
cvs checkout: failed to create lock directory for `/cvsroot/openser/sip-server/modules/imc' (/cvsroot/openser/sip-server/modules/imc/#cvs.lock): Permission denied
cvs checkout: failed to obtain dir lock in repository `/cvsroot/openser/sip-server/modules/imc'
cvs [checkout aborted]: read lock failed - giving up
is imc modules supposed to be in rel_1_1_0?
-- juha
Hello everybody,
we would like to invite all of you to join the OpenSER Summit to be held
during the VoN Autumn Europe, Berlin, November 6-8, 2006. The access is
open for anybody, free of charge. Also, all registered participants will
get free entry passes to the VoN Exhibition. Courtesy of Pulver Media we
can provide full conference passes (exhibition, conference, special
events at VoN) during the summit, but in a limited number, so
first-come, first-served.
Agenda is published at openser site, and will be completed soon with the
panelists and topic names:
http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=…
To register, please fill and submit the form at:
http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=…
or, if more convenient to you, send an email with your details at
team(a)openser.org. Your request will be recorded and you will be
contacted soon for confirmation.
Main focus for this event is to reveal the status of OpenSER project --
achievements and roadmap (board members and main developers of the
project will be there), discuss businesses cases of OpenSER, network
people for business opportunities and cooperations (many executives will
attend the event).
Just to list a few hot topics in the agenda:
- scalability and distribution
- secure communication with SIP using OpenSER
- 3/4G and convergence - fixed mobile convergence and peering
heterogeneous networks
- carrier grade and hosted infrastructures, IMS
- integration with other open source (and not only!!!) VoIP application
to deliver complete and robust VoIP platforms
Looking forward to meet you there.
Daniel Mierla
--
OpenSER Project
http://www.openser.org
this one failed. i didn't check if it is still relevant.
-- juha
---------------------------------------------------------------------
when i look rel_1_1_0 using ViewVC i see core directory mi and imc
module for example. those should not be there:
http://openser.cvs.sourceforge.net/openser/sip-server/?pathrev=rel_1_1_0
this in addition to the earlier problem that anon cvs checkout fails.
-- juha