Hi,
When using the Jabber Gateway, some users get the following error from time to
time:
ERROR: Connection to Jabber server lost. You have to login to Jabber server
again (join again the conferences that you were participating, too).
sip_to_jabber_gateway says:
INFO: Your are now offline in Jabber network. Thank you for using SIP-Jabber
gateway.
Do you know the reason why these messages appear and if it is possible to
avoid them?
Thanks,
Jaime
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Maxim,
Thank you very much for your report.
I'm not the code owner who has the last word, but I agree there is
a memory leak. It seems to me that insert_new_lump family is ok as
long as the calling functions care to free their lump buffers on
failure -- that's where the leak lives.
Which imho happens at few places on CVS:
+insert_new_lump_before
- save_ruri in rr/loose.c
- textops/append_hf_helper.c
- (on contrary, it is ok in maxfwd/add_maxfwd_header, rr functions
calling insert_RR, and msg_translator.c)
+insert_new_lump_after
- search_append_f in textops.c
- replace_f in textops.c
- (ok in build_req_buf...)
The memory leak is unlikely to occur, as it is triggered by lack of
private memory which (unlike shmem) hardly gets exhausted -- so
operation is fortunately not affected. It will be fixed in the next
release.
-Jiri
ps -- I swear on cscope :)
>Folks,
>
>I've noticed that there are multiple potential memory leaks in SER.
>The problem is that if a insert_new_lump*() function returns a NULL
>for some reason (currently the only condition is memory allocation
>error), it doesn't free the memory buffer passed to it and most of
>the code doesn't care to deallocate that buffer after NULL is returned.
>It could be easily fixed and probably needs to before the next version
>is released.
>
>Thanks!
>
>-Maxim
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Goal: To learn about ser in a "non-critical" and very small PBX-like
environment so as to be able to understand nuances of the system in a
production environment at a later date at various firms whose owners
who have expressed to me a high degree of interest in SIP call
routing for larger enterprise and CLEC implementations.
Sub-Goal: to make all calls into and out of my house routed via IP to
alternate destinations based on ser routing configuration. I have
subscribed to a long-distance plan via "iconnecthere.com", I have a
PRI gateway configured at a remote location (for calls into two area
codes only) and my plan is to have a Cisco 2610 with FXO card for
terminating my "local" phone line. My house phones are all Cisco
ATA-186 devices. Based on called number, my calls will be sent to
the iconnecthere.com SIP service, the Cisco PRI gateway, or the Cisco
2610 single-line gateway. Calls will also be routed appropriately
based on number called on an inbound basis from any of those three
gateway systems (PRI, 2610, or iconnecthere.com)
Progress: I have phone-to-phone calling working well, and I have
phone-to-PRI gateway calling working well in both directions. I have
not yet received the 2610, so I do not have the single-line analog
gateway running, but I don't expect any issues with that system, as I
understand the Cisco implementation of inbound/outbound VOIP sessions.
Problem: My "iconnecthere.com" account is a username/password
protected account which (of course) requires a UA at my side of the
connection. To forward calls from the various phones in the house, I
would need to have something re-write my username/password requests
on the fly when they are sent out to the iconnecthere.com SIP
proxies/gateways. My first assumption is that I'll need a (sigh)
B2BUA to act as a gateway, running (for convenience) on the same UNIX
platform as ser. After thinking about it for a while, I am
uncertain if that is required, but at this point I can't determine
what I need to do.
I would appreciate hints on:
a) Wether I need a B2BUA at all, and if not, what config options
should I be looking at in ser?
b) If I do need a B2BUA, what would you recommend? I'm using
OpenBSD as my platform, and (personally) I'd like to stay away from
Java for the moment.
Continuing discussion:
I could see this as a fairly useful toolkit trick for a small
business who wants to replace their phone switch with SER or ser-like
systems. If you've got an office with 10 people, it may make
economic sense to simply get "generic" accounts with a SIP long
distance gateway provider like iconnecthere.com (there are others)
and allocate each of those accounts to individuals in the
organization. This is not a solution for a large-scale operation for
the various reasons outlined in the
http://www.iptel.org/info/trends/#b2bua texts, but the number of
small-scale shops out there is very large and a simply understood
package (and simply billed) is what would be desired for many places
who still use under ~10 analog lines for outbound dialing from their
PBX. Any outbound calls to certain prefixes would always be pushed
through a specific account for LD calling.
Additionally, inbound calling through a similar service would have
to also come in via the same mechanism, with a REGISTER request being
sent by the B2BUA and all subsequent calls being routed through the
SER proxy.
PS: I'd appreciate any open-source hints on how to get an ATA-186
(v2.15) running behind NAT with ser on the "outside" of the NAT,
without statically configuring the "NAT" address on the ATA-186 every
time the outside address changes. Lots of keyword matches found on
Google, but very few clues to be scraped from the resulting documents
as to "how" do it from the server side.
JT
Greetings to all and happy new year
I'm in the process of installing ser 0.8.10 on Linux
(intel) and when I run Too much shared memory the
"ser"executable I receive the following error message
"Too much shared memory demanded: 33554432"
When I turn the debug option on I receive the
following:
0(4019) WARNING: hash function optimized for 1024
entries
0(4019) WARNING: use of 65536 entries may lead to
unflat distribution
0(4019) ERROR: shm_mem_init: could not attach shared
memory segment: Invalid argument
0(4019) could not initialize shared memory pool,
exiting...
Too much shared memory demanded: 33554432
Any ideas?
Thanks
__________________________________________________
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Dear Sir/Madam,
We are developing a sip stack.
We wanted to know whether you can provide any sip server for testing our stack.
We also wanted to know the procedures invloved in it.
Thanks in advance
Geetha
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Dear Sirs,
I am having some strange problems when trying to use b2bua for
accounting and call duration limiting with SER proxy server. The idea
is simple: since the SER can route SIP messages depending on their
source address, we can force incoming SIP messages to be passed to
B2BUA for accounting purposes first, and after the same request
re-enters the proxy from B2BUA pass it to the final destination. Call
flow looks like the following:
-----
|UA2|
-----
^
|
|4
|
----- 3 -------
----- 1 | |<-----------| |
|UA1|---->|SER| 2 |B2BUA|
----- | |----------->| |
----- -------
For some reason, it doesn't work in such configuration. The problem is
that B2BUA's UAC keeps resending `200 OK' replies ingnoring ACKs it
receives from UA2 until timeout hits, after which it considers the
call dead, despite the fact that both UA1 and UA2 think that the call
is established. Maybe it has something to do with the fact that it
sends to and receives messages from the same host (SER), but I don't
think that this should be a problem, since those two call legs have
different call id's, so that b2bua should be able to distinguish
between them easily. Attached please find tcpdump logs of one such
session, here 192.168.1.1 is UA1 (originating), 192.168.0.9 is UA2,
192.168.1.100 is the host running both SER and B2BUA (the former uses
port 5060, while the latter - 5065). There are two files:
ser-b2bua.log is the log of udp exchange between SER and B2BUA and
ser-ua1ua2.log - log of exchange between SER and UA1/UA2.
Any ideas are appreciated.
Thanks!
-Maxim
On Mon, Jan 20, 2003 at 03:22:33PM +0100, Jiri Kuthan wrote:
> At 09:53 PM 1/13/2003, you wrote:
> >Attached please find the patch, which implements t_on_positive()
> >approach. Now my config looks like the following and everything now
> >works as expected:
>
> thanks for the patch -- I will try to integrate it as I move ahead
> with other changes to TM. Does your nathelper module work now?
Attached please find more fresh patch with several bugs smashed. It
is probably the last one on this topic, because it works reasonably
well and we now have moved to adjusting b2bua for our needs (mostly
radius auth/acc). In addition to nathelper module, this patch
implements rport support as specified in the correstponding IETF
draft.
Thanks!
-Maxim