Hi Brett,
I think using script variables should work -
http://www.opensips.org/Resources/DocsCoreVar#varscript.
Regards,
Anca
On 10/13/2010 01:31 AM, Brett Woollum wrote:
Hi Anca,
I figured out the solution that you were referring to. I added the
local_route section and changed the header
Hi,
stun module allows you to run a stun server on the opensips platform...
so your device sith the stun client uses your proxy ip(s) to access a
stun server.
Paul
CheeWii wrote:
Hi,
I want to solve the nat traversal problem, and I read the great
book Building Telephony System with
I'm not explicitly routing that BYE, but as Andrew Pogrebennyk pointed
out, since I'm rewriting the Contact header with OpenSIPS address it
is expected that BYE won't go any further than OpenSIPs proxy and
therefore my only solutions seems to be the configuration of the b2bua
modules. Isn't it?
I'll check it later, but the fact is that I have to solve first the
problem that prevents the INVITEs generated by OpenSIPs to be sent to
the internal jain slee server.
I have confirmed that after the call to ds_select(), the value of $du
points to such server (which has been obtained from the
Hello,
sorry for the newbie stupid question.
Is it normal that after a loose_route() the $du is set to null?
this is my config:
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
Hi,
Sorry for the newbie question. How do you change the port number in
opensips? I want to change it from 5060 to something else however when I do
so on opensips.cfg and restart opensips, Opensips doesn't seem to listen on
the port I have specified. Do I need to change the number elsewhere?
I'm not sure that process persistence is what Brett was looking for but rather
Dialog persistence. I have found that local memcache support is very fast and
takes care of this type of need quite well.
Using a unique key, made up perhaps of the SIP call-id and type of value like
I was forced to remove the OpenSIPS b2bua as it seems it was causing the
origination gateway to choke on these 200 OKs. Dialogs are no longer
hanging, but I still see errors on the origination gateway that are
concerning:
/usr/local/sbin/opensips[4399]: ERROR:core:parse_from_header: bad from
Hello all,
The current implementation of the pua module allows sending custom
headers in PUBLISH requests.
I would like to have the ability to save those headers in the presence
server and relay them to presence subscribers via notifications
(NOTIFY requests).
This would require adding a new
Hi guys,
Am a SIP and Opensips newbie and am trying to create a proof of concept on a
unique (I think) Opensips and SIPS implementation. Basically am developing a
web-based SIP Phone based on PHP and using Opensips. Because of the nature
of the application, users will have the same domain and
Over the past couple weeks I have been getting occasional segfaults just
prior to (or perhaps in the process of) a t_relay() in my failure_route.
Still haven't gotten to the bottom of the root cause in my config, but I
was able to find and fix the symptomatic code in the
pre_print_uac_request()
I could be wrong but does your phone register with the proxy and then the
proxy create a location record for each user? That would be how opensips
would know how to route it.
On Oct 13, 2010 1:04 PM, James Mbuthia jmmbut...@gmail.com wrote:
Hi guys,
Am a SIP and Opensips newbie and am trying
Hi Dimitri,
I don;t think it will pay off - especially that you can do it from
script level (multi checks in if statement)
Regards,
Bogdan
DM wrote:
Hi Bogdan,
It is hard to implement with multiple codecs?, eg. if I store it like
this:
$avp(s:codlist)=PCMA|G723|tel;
So not with just
David,
In this case, instead of using the expensive solution of a b2bua, why
don't you take care of properly re-write the contact. The question is
why do you replace the received contact with the IP of your opensips ?
Regards,
Bogdan
David Santiago wrote:
I'm not explicitly routing that BYE,
Thanks Logan,
The patch was already uploaded on SVN trunk and stable.
Regards,
Bogdan
logan wrote:
Bogdan, I've applied the patch you provided and initial testing has
been successful. I'll keep you posted on the rest of my tests. Thank
you for your attention to this!
Hi Gabriel,
why don't you use in DR 2 different rule with different strip and
pri_prefix like:
rule1:
matches:00593
strips: 5 digits
pri_prefix: 7424
rule2:
matches:001 (or you can use default rule with empty matching prefix)
strips: 0 digits
pri_prefix: 7424
The GWs
Hi,
regarding the decode_uri issue : the uac module stores a cookie in the
RR header, so at Route time it will be able to restore the original FROM
/ TO hdr. I suspect that the one of the parties (caller or callee) is
messing around with this RR parameter (vsf , vst), most probably with
the
Again, can you post a trace of the message causing the error?
Regards,
Bogdan
thrillerbee wrote:
I was forced to remove the OpenSIPS b2bua as it seems it was causing
the origination gateway to choke on these 200 OKs. Dialogs are no
longer hanging, but I still see errors on the origination
Hi Leon,
Have you created the ocp_admin_privileges table ? see the install
instructions for OpenSIPS control panel on
http://opensips-cp.sourceforge.net/
Regards,
Bogdan
Leon Li wrote:
Hi ,
I followed the instruction in “build telephony system with opensips
1.6” to install opensips-cp.
Hi Stefano,
yes, it is possible - if the request does not have any other Route hdr
to indicate the next hope, routing based on RURI will be done, so the
loose_route function will not set any $du (destination URI) - this du
is actually a kind of outbound proxy used by opensips when it wants to
Hi James,
how do you set the port? via port param or via the listen param? are
you sure you restart was effective?
Regards,
Bogdan
James Mbuthia wrote:
Hi,
Sorry for the newbie question. How do you change the port number in
opensips? I want to change it from 5060 to something else however
Hi Dave,
In failure route, how do you add the new destination/branch ?
Also, do you have a branch route set ?
Regards,
Bogdan
Dave May wrote:
Over the past couple weeks I have been getting occasional segfaults just
prior to (or perhaps in the process of) a t_relay() in my failure_route.
Hello Everyone,
In an attempt to figure out the best way to build my OpenSIPS config with the
B2BUA module included, I've started over with a very simple script implementing
nothing but the B2BUA module (and usrloc). My goal is to allow the phones to
place calls between them and transfer the
Hi Marcio,
The answer is:
Oct 13 17:05:47 perseu /sbin/opensips[13077]: DBG:dialplan:build_rule:
references:1 , max:1
Oct 13 17:05:47 perseu /sbin/opensips[13077]: DBG:dialplan:destroy_rule:
destroying rule with priority 1
It looks like opensips rejects the rules while loading them at
Unfortunately the mailing list won't let me send replies with my logs or config
file since they're too big.
You can find them on this page: http://www.woollum.com/temp/opensips.txt
Thanks!
Brett Woollum
br...@woollum.com
- Original Message -
From: Brett Woollum
Hi Taisto,
Your problem is not timer related or how serial forking is done in
opensips (I will comment on these in a later reply).
Right now, the quick answer to fix your problem: failure route must be
re-armed after each branch - this is why your failure route does not
catches the end of the
Hello.
I am planning to provide opensips with a kind of mechanism to manage
customer services/features like call-forward/VM/follow-me and so on.
It should work in following way: If $rU is provided in subscriber
table then user enabled service name is obtained from some db table.
On the basis of
But when does the proxy create a location record?
When the callee phone registers it gets an authenticate challenge and
after the challenge it gets a 200 Ok header.
When the caller phone calls it gets a 407 proxy challenge and after
verification gets a 100 response.
According to my
Hi Bogdan,
I set the port on opensips.cfg like:
port=5060
/* uncomment and configure the following line if you want opensips to
bind on a specific interface/port/proto (default bind on all available) */
listen=udp:198.162.1.0:5060
I change the 5060 to another number then restart using
Yes, I do set a branch_route to strip RPID header and strip/set the
P-Asserted-Identity header.
I'm using drouting, and that portion of the failure_route is based on
the code from Flavio's book (with the addition of e164 mangling,
t_on_branch to handle the header stuff mentioned above, and
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