Hello,
I am now working with iptel express router source code ver 0.8.10
on a Suse 8.0 machine with gcc version 3.2
How can i send a signal to the server for shutdown and deinstall the
installed server later which ist installed from a
1) tar.gz source with make install
and if different
2) tar.gz source by unzipping the code under directory "/"
Thanks
Hi folks,
( I think!) I have successfully installed SER server with the ser-mysql and
ser-jabber packages ( I have also a mysql server and a Jabber server
running).
For testing I have established a SIP IM session between two MS Messengers
and it did work.
After this I have tried to establish a IM session between a user SIP (MS
Messenger) and a user Jabber (Exodus), but when the messenger sends a
message for the jabber gateway the following error occurs:
"ERROR: Your message was not sent. You do not have permision to use the
gateway."
What could be the problem here?
Best regards,
Toni
I have installed IpTel on a SuSe 8.0 machine and have 2 Windows machines
attached with a hub.
When i try to add a user with: serctl add username ....-
then i have a problem executing mysql.
So is it because of the wrong installation or do i have to install the
mysql packet from the SuSe control center?
Thanks in advance.
I also have found myself moving over to Free- and OpenBSD for most
workstation and server needs, but I keep an install of Mandrake handy
for those few apps that don't port to well over to the BSD platforms.
What I'm really interested in is stund support in OpenBSD that
integrated w/ pf, seeing as how the majority of my firewalls are now
OpenBSD 3.1-3.2 boxes running the new pf code. Ser itself runs great on
FreeBSD 4.7-RELEASE, which is what I use as my testing platform, but
this NAT stuff is really killing me when it comes to bringing in
potential callers over the WAN. Any word on when a *BSD-compatible
binary of stund might be available? Thanks.
--
Art Mason CCNA
Network Consultant
Maverick Datacom
E-mail: amason(a)maverickdatacom.com
Folks,
I discovered that the tm module generates non RFC3261 compliant branch
id when forwarding ACK to the final 200 OK response. The RFC, to which
ser claims to complain by prefixing generated branch id with magic
"z9hG4bK", mandates that ACK to the final 200 OK response should carry
on the same branch id as that response and the initial INVITE. On contrary,
ser currently adds RFC3261-compliant branch ids for everything, but
that final ACK, which in default configuration is being sent with branch
id of `0'. This creates a compatibility problem when the next hop uses
that branch id for matching 200 OKs to subsequent ACKs, example of such
UA is Vovida's b2bua software.
I do not have a fix right now, as I am away from my lab, but it should
be pretty trivial to fix by pre-loading msg->add_to_branch_s with a
correct value in tm module before calling forward() and non-overriding it
in the forward() if it is already non-NULL.
-Maxim
Hello,
Above all my questions before, i would like to make everything clear from
the beginning.
I was trying to IPTEL express router under the directory which came after
unzipping ser-0.8.10_src.tar.gz with the command "tar -xzvf"
on my SuSe-Linux 8.0 machine with kernel 2.4.18-4GB .
I got some errors during compilation. Is this version 0.8.10 compatible
with
SuSe 8.0? When not with which SuSe Linux distributions is it working
properly?
Could you send me compilation and installation procedures or refer to the
related files if possible?
Waiting for your reply,
Best regards,
Hi *,
we have a 3PCC-scenario and we want to modify an existing session.
(Change IP and Port an UA is listening to)
In RFC 3261 section 14 is described how to modify existing sessions
throught an re-Invite but SER sends me an OK without sending the 2nd
Invite to the UAC.
Can you imagine what I have done wrong?
(My scenario acts like Flow II in draft-ietf-sipping-3pcc-02.txt,
becourse my UAC is buggy and gives me to to an Invite without SDP a 500)
(The Screenshots of my test are attached)
Thank you!
Karsten
Daniel,
Not sure if it has been asked before.
Is presence support from SIP to Jabber and viceversa completely working in cvs
version?
Regards,
Jaime
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