Hi list,
I am running a NATing proxy. Everything works fine, except that
some of my neighbour proxies send me BYE requests with an incorrect
Route header containing their own IP address.
The result is naturally that my proxy sends back the request and
the call is never ended.
I know this is a wrong behaviour and I have contacted the remote
administrator to have him fix his proxy. But in the meantime
I would like to find a workaround for my users.
I tried to do the following, before calling loose_routing(). I am
not sure if it's relevant or not nevertheless:
% if (method == "BYE") {
% # XXX Remove buggy Route header from BYE packets.
% xlog("L_ALERT", "Removing buggy Route header from BYE request");
% remove_hf("Route");
% lookup("location");
% }
However, I raised the verbosity level and found that the Route header
is read as soon as the packet is received, therefore the above
block is not really useful.
Thank you for your ideas.
Regards,
--
Jeremie Le Hen
< jeremie at le-hen dot org >< ttz at chchile dot org >
Hello list.
Sorry to ask this question here, but i can't find the answer in the RFC3261
Is this "VIA" permited according to the RFC?
Via: SIP/2.0/UDP 69.20.6.205:15060, SIP/2.0/UDP 69.1.229.69:4081;branch=z9hG4bK-58e9ccfcef0f4c99915b8007cb9713e1-1156793814
It seems to be two VIA's in ONE line...
Thanks
Regards,
Ricardo.-
Hi all, I have tried implementing SIP redirection as mentioned in the example script below:
route{
# for testing purposes, simply okay all REGISTERs
if (method=="REGISTER") {
log("REGISTER");
sl_send_reply("200", "ok");
break;
};
# rewrite current URI, which is always part of destination ser
rewriteuri("sip:parallel@iptel.org:9");
# append one more URI to the destination ser
append_branch("sip:redirect@iptel.org:9");
# redirect now
sl_send_reply("300", "Redirect");
}
But that does redirection only for that account as the account is included in the code. I need the code to be able to redirect any call. Can you tell me how I can generalize the code so that
redirection works for any account I choose to redirect?
Thanks in advance,
Padmaja
Hey,
are there working ways for authentication with sip trunks? i've read cisco
gateways (PGW) and cisco call manager can do this.
anyone knows if openser+asterisk support it too?
Hi!
I have registered, thanks to openserctl, an
user :'sip:testeur@192.168.0.102'. But when I tyr to reach him, the
transaction isn't considered as local processed one. In location table
there is:
+username + domain + contact +
|testeur | | sip:testeur@192.168.0.120:5080 |
+---------+--------+--------------------------------+
-192.168.0.120 is the IP address of the user's phone.
-192.168.0.102 is the openser IP address.
The openser.cfg is as the initial openser.cfg.
Thanks,
Michel
Hello OpenSer Users,
It seems that many of you are also Asterisk users or at least subscribe
to asterisk-users. Some of you might recognize AstLinux...
Just in case you don't:
http://www.astlinux.org
---and---
http://www.sf.net/projects/astlinux [in transition]
Anyways, I have long hoped to build a version of AstLinux that includes
OpenSer. Basically, what I have done is taken astlinux-trunk, add
openser, add rtpproxy, de-select the Asterisk packages, and build!
What I got was 11mb (gzipped) disk images ready to be written to CF
cards/hard disks. As usual, I have created two images:
*net4801.img.gz is for the Soekris net4801 SBC
*i586.img.gz is for just about any i586 and higher machine
There is also an ISO image ready to boot on or install to most i586 or
higher machines. You guessed it, it is the one with .iso in the
filename. It includes both the net4801 and i586 images.
These are 48mb gzipped disk images, with GRUB bootloader and ext2 root
filesystem. If you want to write them to disk, try this:
gunzip [image.gz]
dd if=[output file.img] of=/dev/sdb # <--- VERIFY THIS DEVICE
Where /dev/sdb is where your USB CF flash adapter is.
If you want access to the ext2 filesystem, try this:
gunzip [image.gz]
mkdir /mnt/image
mount -t ext2 -o loop,offset=32256 [image.gz] /mnt/image
If you are curious about how these images were created, take a look at
SVN on the astlinux project page.
The OpenSer version used for these builds was openser-1.1.0-notls. I
built every module that did not depend on an external library that I
don't have yet (sorry, no mysql, jabber, or radius yet).
I need help!!!
For the time being, I don't have mysql. Also, because AstLinux is
targeted for compact flash, mysql isn't a good idea anyways. I was not
able to get a working configuration in the time alloted using DBTEXT.
Does anyone have a sample config that I can put in this build? As of
now, openser, rtpproxy, openser modules, and all of the astlinux-style
init scripts are done. All I need is a good sample config file to make
it "plug and play".
I also need testing!
My current to-do list:
- add mysql (and the openser modules, of course)
- add radius support (freeradius?)
- jabber/xmpp (depending on size, complexity, etc)
- fix astshape (iproute2-based traffic shaper) RTP TOS values to be the
same that rtpproxy uses, and standardize this across all astlinux builds
- COME UP WITH A NAME. I have no idea what to call this thing. Any
suggestions? SERlinux (or something like it) is too obvious and boring!
You can grab the goodies here:
http://www.krisk.org/openser/
Regardless, I hope that someone finds this interesting or useful. Let
me know what you think!
--
Kristian Kielhofner
Hi,
(just a little bit on the side-bar)
I've had some discussion with the developer of twinkle (which is, BTW, a
very nice linux softphone) regarding handeling icmp port unreachable and
switching to a lower priority SIP router (by DNS SRV).
It seems like the dist I'm using (Dapper, kernel 2.6.15-26) will not
easily allow the applications to receive the icmp on an unconnected udp
socket, hence failover will not work. However on an 2.6.11.4-21.11
kernel it works.
Does anyone know if there are any kernel opts which should be added for
having this kind of support?
Just my thoughts this morning......
Hi,
Hi, im using an Openser integrated with Asterisk to route PSTN calls,
this is working great, but i also want to route P2P calls to Asterisk
like this
UA1 -> Openser -> Asterisk (calls DIAL(SIP/xxxxx@openser) -> Openser ->
UA2, to use its mediaproxy, and account all calls.
When UA1 calls UA2, everything goes fine, UA2 receive the INVITE and
sends the OK to UA1, the only thing is that UA2 waits for an ACK from
the OK that never comes.
Have someone tried to do this? How can i configure openser and asterisk
for this?
Thanks
Lucio Maciel
greetings,
I have digest and basic auth setup. I am trying to figure
out how to setup IP auth. I have
read a lot of documentation but I can't seem to find anything to help me
understand how to accomplish
this. Can someone point me in a direction to some information on how to set
this up?
Brian
