Hello all,
In order to check that our Openser is running properly I use sipsak
periodically. The command is :
sipsak -v -N -U -I -s sip:sipsak@213.5.xxx.xxx:5060 -a xxxxxxx
This works fine 99% of the time, however sometimes I get the following
from sipsak :
warning: ignoring -i option when in usrloc mode
received:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP
127.0.0.1:54393;branch=z9hG4bK.4d383307;rport=54393;alias;received=213.5
.43.131
From: sip:sipsak@213.5.43.131:5060;tag=44d4f5ef
To:
sip:sipsak@213.5.43.131:5060;tag=97e44c910458bd56beb40ca4028d7cc8.91db
Call-ID: 1154807279(a)127.0.0.1
CSeq: 2 REGISTER
P-Registrar-Error: Invalid CSeq number
Server: Altec Telecoms SIP Proxy
Content-Length: 0
error: didn't received '200 OK' on register (see above). aborting SIP
failure
sipsak is running on the same server with Openser. Since the CSeq is 2
it means that Openser has received the first REGISTER, has asked for
authentication, sipsak replied, but then Openser says "Invalid CSeq
number". Has anyone seen this happen before and/or has a reasonable
explanation?
thank you
George
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If I am not using a failure_route to catch failed responses to
transactional events, but have an onreply_route, does that guarantee
that *all* failures will be caught by the onreply_route without being
diverted and muffled somewhere?
Based on the behaviour I am witnessing, it seems that is correct, but
I wanted to verify.
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
Direct : +1-678-954-0671
Hi, for now, the only way to test flags is by $bf (not useful if there are
enabled some bflags) or doing something as:
if (isbflagset(6)) {
$var(bflag6) = 1;
}
xlog("L_INFO","--- $var(bflag6) ----\n");
As a suggestion, couldn't be a way to access to the value of a flag like:
xlog("L_INFO","--- $bf[6] ----\n");
This could be useful to when matching flgas with table fields and so.
Regards.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Hi, I'm wondering which solution (A or B) is more efficient in case method is
NOT an INVITE:
A) if ( (method=="INVITE") && (isflagset(5) || isbflagset(6)) )
B) if ( mehtod=="INVITE" )
My question is: Is faster B because does less comparasions?
or does OpenSer break A conditional during first condition (method=="INVITE")
because it's false so then A and B are equally efficient?
And which is more efficient (C or D) when all the conditions are true?
C) if ( (method=="INVITE") && (isflagset(5) || isbflagset(6)) ) {
....
D) if ( method=="INVITE" ) {
if ( (isflagset(5) || isbflagset(6)) ) {
...
Thanks for any explanation.
Best regards.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Hi, I've a multidomain OpenSer and I want to allow REFER (with authenticacion
before) just in calls between users of same domain, but it's quite complex:
- userA(a)domainA.com INVITE's userB(a)domainB.com.
- During the call userA(a)domainA.com does an in-dialog REFER.
Because the REFER is in-dialog the message is like:
--------------------
REFER sip:userB@80.98.123.23:5060 SIP/2.0
From: sip:userA@domainA.com
To: userB(a)domainB.com
Refer-To: ...
...
--------------------
How can my OpenSer know which domain this REFER goes? of course the "To"
header is not valid at all.
Could be a solution a SQL query to "location" table looking for the URI
(sip:userB@80.98.123.23:5060) and getting the username and domain of this?
Any other option?
Thanks a lot.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Hi, in order to not apply RtpPrxy if the called is an Asterisk (that
supports "comedia") I load "usr_preferences" table looking for "pbx_comedia"
attribute for called domain.
If it's, then it's created a avp:
avp(s:pbx_comedia)="yes".
But when I try to test this AVP in the "on_reply_route" (to not apply
RtpProxy) this AVP has null value.
In fact it's easy to see:
------------------------------------------------------
route[3] { ; INVITE route
$avp(s:test)="hello";
...
route(1);
}
onreply_route[1] {
xlog("L_INFO","----- avp(s:test) = $avp(s:test) ------\n");
}
------------------------------------------------------
The xlog of onreply_route shows:
----- avp(s_test) = null -----
is it normal? aren't AVP's transaction aware?
Thanks.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Hi there!
I'm using ser0.9.6 to do the call transfer, which part of the configuration
that I need to configure in ser.cfg? Do I need to create a new route?
Thanks.
Cheers,
Roa Yu :)
-----Original Message-----
From: serusers-bounces(a)lists.iptel.org
[mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu
Sent: Friday, October 12, 2007 2:45 PM
To: kgfleischmann(a)t-online.de
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer
Hi Klaus!
The UA does not received the REFER sent by SPA942. What should I need to
configure to ensure that the UA receive the REFER from the transfer
agent(SPA942)?
Thanks.
Cheers,
Roa Yu
-----Original Message-----
From: serusers-bounces(a)lists.iptel.org
[mailto:serusers-bounces@lists.iptel.org] On Behalf Of
kgfleischmann(a)t-online.de
Sent: Friday, October 12, 2007 2:22 PM
To: serusers(a)lists.iptel.org
Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer
The NOTIFY is always sent from the UA, which received the REFER. SER
only relays it. Did you check this?
Cheers
Klaus
roayu wrote:
> Hi there!
>
> After SPA942(transfer agent) sent REFER to SER, SER does not send a NOTIFY
> to SPA942(transfer agent). What's causes this? Can anyone tell me.
>
> Thanks.
>
> Cheers,
> Roa Yu :)
>
> -----Original Message-----
> From: serusers-bounces(a)lists.iptel.org
> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu
> Sent: Thursday, October 11, 2007 4:34 PM
> To: Atle Samuelsen
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer
>
>
> Hi Atle,
>
> Let me make my scenario clear on you.
>
> I have 2 softphone and 1 IP phone(SPA942). I would like to call from
> Softphone A to SPA942, then use SPA942 transfer the call to Softphone B.
> But, when I do so, the connection still remained on between SPA942 and
> Softphone B.
>
> I managed to get Softphone A onHold and the SPA942 managed to call to
> Softphone B. Once I press on the 'xfer' button after calling Softphone B,
A
> still onHold and no SIP signal to unHold A and to terminate B.
>
> If I would like to configure the ser.cfg file, which part that I need to
> configure? And how do I do that?
>
> Thanks.
>
> Cheers,
> Roa Yu
>
>
> -----Original Message-----
> From: Atle Samuelsen [mailto:clona@cyberhouse.no]
> Sent: Thursday, October 11, 2007 3:19 PM
> To: roayu
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer
>
>
> Hi Roa,i
> * roayu <roayu(a)ctisys.net> [071011 04:01]:
>> Hi, there!
>>
>> I have some questions and help.
>> 1) I would like to know can call transfer being done when the
>> connection is P2P? When I tried to use relay (mediaproxy) to do the call
>> transfer, it's able to transfer the call to the other party whereas when
> the
>> condition is P2P, it just can't pass the correct signal.
>
> I'm proberbly only tierd (9:15 am here) but P2P? Can you provide the
> signalling and ser.cfg so we can understand what you are trying to
> establish?
>
> I personally think you are trying to do a call transfer from one ua to a
> other, where the original call went true mediaproxy, but you do not want
> the "new" transferd call to go true it.
>
>> 2) Can SER support SPA942 ? Or is there some other settings that I need
>> to configure on SER ?
>
> SPA942, SPA962, SPA2102, yea.. all Linksys SPA products work like a
> charm with SER (Who has a 942 as his primary phone these days)
>
>
> Best Regards
>
> ATle
>> Thanks.
>>
>> Cheers,
>> Roa Yu
>>
>> _______________________________________________
>> Serusers mailing list
>> Serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
> _______________________________________________
> Serusers mailing list
> Serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
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>
>
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