Hi,
Does any one know of a useful way of getting(turning on) SER debug report. I am trying to locate where my call is failing with the SER and currently use the "ngrep" but it does not seem to reveal much useful internal report. most of it looks external similar to ethereal.
Will appreciate any share of info in this regard.
Cyp.
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Hi Tim,
As I suspected, it is empty branch is triggered by a bogus SIP flow. The
ACK generated by the caller device is bogus as the TO uri is different
than the TO in INVITE.
Also the caller device does use branch at all in VIA and forces openser
to do pre-RFC3261 transaction matching with involves the TO uri.
As the log show, the ACK does not match any transaction:
DEBUG: t_lookup_request: start searching: hash=11223, isACK=1
DEBUG: proceeding to pre-RFC3261 transaction matching
DEBUG: t_lookup_request: no transaction found
So, the ACK is forwarded as stateless request - for this case, there is
complicated alg for computing the branch param based on the branch
params from the previous VIA headers. But the previous VIA is bogus (no
branch at all) and openser finds itself in the impossibility to compute
a branch for its VIA.
regards,
bogdan
Tim Madorma wrote:
> Hi Bogdan,
>
> Attached is the log and the openser.cfg file.
>
> thanks,
> Tim
>
> On 5/30/07, Tim Madorma <tmadorma(a)gmail.com> wrote:
>> Hi Bogdan,
>>
>> I have attached an html file that shows the call trace (note that the
>> OpenSER is 65.185.232.62:5061in the flow). I will send the log file
>> and openser.cfg in a separate email since it would put me over the 40K
>> limit.
>>
>> thanks,
>> Tim
>>
>> On 5/30/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
>> > Hi Tim,
>> >
>> > the VIA branch param must not be 0, indeed. Could you provide an
>> network
>> > trace for the entire call and the debug log for level 9?
>> >
>> > regards,
>> > bogdan
>> >
>> > Tim Madorma wrote:
>> > > Hi,
>> > >
>> > > In testing with OpenSER 1.2 I noticed that the branch parameter
>> of the
>> > > Via in the ACK to 200 OK has a value of zero. This does not seem
>> > > right. From 16.6 of RFC-3261:
>> > >
>> > > The proxy MUST insert a Via header field value into the copy
>> > > before the existing Via header field values. The
>> construction
>> > > of this value follows the same guidelines of Section
>> 8.1.1.7.
>> > > This implies that the proxy will compute its own branch
>> > > parameter, which will be globally unique for that branch,
>> and
>> > > contain the requisite magic cookie.
>> > >
>> > > Please advise.
>> > >
>> > > thanks,
>> > > Tim
>> > >
>> > > _______________________________________________
>> > > Users mailing list
>> > > Users(a)openser.org
>> > > http://openser.org/cgi-bin/mailman/listinfo/users
>> > >
>> >
>> >
>>
>>
I have set up an OpenSER to run on port 5062 instead of 5060. Attached
to OpenSER I have 2 videoconferencing clients which communicate via SIP.
What's working so far
- internal client (registered on OpenSER) to another internal client
(registered on OpenSER)
- internal client (registered on OpenSER) to another external client
- this all works with the OpenSER and other clients behind their NAT, as
Mediaproxy is installed
What is actually not working is
- another external client connetcing to an internal client (registered
on OpenSER) because I have to adress the the To:URI as
sip.mydomain.com:5062.
- Whenever an external INVITE comed in, it ends up with "message too
big" because OpenSER is cycling through its routing plan until it adds
that many via headers that the message increases 2kb.
- what I see is that openser does not try to find the client on
sip.mydomain.com, instead I feel that it tries to find the client on
sip.mydomain.com:5062 as uri.
- as my operating system (debian etch) cannot resolve this domain, this
runs through the loop
My question:
- is the a way to modify the destinaton header ($tu) that the port is
cut off (name@sip.mydomain.com:5062 ==> name(a)sip.mydomain.com)?
Best regards
peter
Hi,
I could get the blind (unconditional) call forward option working with the
configuration that is provided in the IPTEL website but using the same
config file and the ideas provided I couldn't get the call forward on busy
and no answer working. I get the following error in the /var/log/messages:
May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: mk_proxy: could
not resolve hostname: "ser.iptel.org"
May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: uri2proxy: bad
host name in URI <sip:1234@ser.iptel.org>
May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR:
t_forward_nonack: failure to add branches
May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: w_t_relay
(failure mode): forwarding failed
May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: error:
mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: sl_reply_error
used: I'm terribly sorry, server error occurred (1/SL)
It would be of great help if anybody provide me input on how to resolve this
issue or a configuration file using which I can get the call forward on busy
and no response working. BTW, I am using the ser verison: 0.9.6. I have
populated an entry for call forward no answer/busy in the mysql database for
the table "usr_preferences". Thanks in advance.
Regards,
Ranganath B
Hi,
In testing with OpenSER 1.2 I noticed that the branch parameter of the
Via in the ACK to 200 OK has a value of zero. This does not seem
right. From 16.6 of RFC-3261:
The proxy MUST insert a Via header field value into the copy
before the existing Via header field values. The construction
of this value follows the same guidelines of Section 8.1.1.7.
This implies that the proxy will compute its own branch
parameter, which will be globally unique for that branch, and
contain the requisite magic cookie.
Please advise.
thanks,
Tim
Hi,
I downloaded the source and compiled SER 2.0.0 RC 1 as per instructions on
http://www.iptel.org/ser_2_0_0_release_candidate_1
I get an error when try to check it with 'sercmd' and also can't see the ser
process running
Rgrds,
Umer
[root@umer ~]# /usr/local/sbin/ser
Listening on
udp: 192.168.1.103 [192.168.1.103]:5060
tcp: 192.168.1.103 [192.168.1.103]:5060
Aliases:
tcp: phone03.tb.jasomi.com:5060
udp: phone03.tb.jasomi.com:5060
*: umer-sip.jasomi.com:*
[root@umer ~]# /usr/local/sbin/sercmd
ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory
[2]
[root@umer ~]# ps -A | grep ser
135 ? 00:00:00 kseriod
4931 ? 00:00:00 gam_server
Hi,
I have the following situation.
I have one SER instance with public IP and LAN working as a boundary proxy.
(its only role is to filter the traffic and authorize users)
on the LAN side I have another SER instance with works as registrar and local sip proxy.
How can I force the second proxy this in the LAN network to route the messages to the boundary proxy(no matter what transport is used). The best solution would be to route them only when the calls are made to the users which are localized outside the local LAN on the public NET.
Thank You
In advanced
Cheers Tomasz
