I just finished configuring and testing the project tracker hosted on
SorceForge. My apologies for the "un-explained" emails on the devel list :)
The purpose of the tracker will be of course to have a centralized
system to keep trace of :
1) bug reports
2) support requests
3) feature requests
4) documentation reports
5) patch submission
If needed, other sections can be added. All developer have admin rights
on this tracker (to create new categories or groups, etc).
let's use it and start registering all known pending issues (discussed
so far on the mailing list) in order to avoid forgetting them and to
have a big picture of the current status.
All changes regarding the status of the register items will be emailed
to the devel mailing list.
There are couple of items already registered by me - all project
developers: please review them and update them accordingly if necessary!
For example, for the enum patch, I think Juha will be the best person to
assign it to.
Thanks for the info guys! It sounds like I need to do a little reading
up on cseq to determine if this will even work, or find a PSTN gateway
provider that supports Session-Timers.
On 10/27/05, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
> by sending re-INVITEs from the middle of the path you will increase the
> cseq number differently on each side...so you will need to synchronize
> the cseq value when some in-the-dialog requests are passing through your
> proxy....and that's quite complicated....
> Daryl Sanders wrote:
> >Would it be possible to fake a REINVITE then check the response to
> >determine if a call is still in session? Just brainstorming...
> >- Daryl
I am trying to develop some features which requires to call load_gws
and next_gw in a failure route. For example, in normal routing block,
it forwards the call to a subscriber's ip phone locations. If nobody
answer, it tries to call a pstn number. In the failure route, i called
load_gws and next_gw. However it generates an error "next_gw(): No
ruri_user AVP". Looking at the code, there seems to be an assumption
that first time call to next_gw always happens in the normal routing
block. But in my case, it calls next_gw first time in a failure route.
I even tried to manually add the new uri's username to avp i:1402. It
still won't work.
Is it possible to call next_gw first time in a failure route?
Following that logic should I not be able to just replace the display
and leave the uri as is with...
I have of coarse already trid that with no success.
On Oct 28, 2005, at 3:38 PM, Norman Brandinger wrote:
> If you break your callerid into 2 parts, you should be able to use
> something like the following:
> AVP display: "John Doe"
> AVP uri: "sip:firstname.lastname@example.org"
> The above information has been documented at http://www.openser.org/
> Brandon Price wrote:
>> We are storing cnam (callerid name) values in a mysql database for
>> use with openser. I have openser pulling the values from the
>> database using avpops but I need to know how to insert the value
>> into the from so that it will be passed to my sip user agents.
>> Related code posed below
>> modparam("avpops", "avp_aliases", "cnam=s:cnam")
>> avp_db_load("$from/cnam_scheme","cnam/$cnam_scheme"); #sample
>> query -- select cnam from cnam_cache where phonenumber='13143212222'
>> this should now store that value in $cnam.
>> now how do I put that value for example "John Smith" in the from
>> portion of sip messaging so that it displays on the users sip
>> phone, or phone attached to an ata?
>> Users mailing list
---------- Forwarded message ----------
From: Rehan Ahmed <rehan33(a)gmail.com>
Date: Nov 1, 2005 8:02 AM
Subject: DIDX.org - DIDXchange
I wanted to know if someone here is using did numbers from DIDX.org and
able to make them work on SER yet.
After I installed the latest CVS version, I found SER could't be
started by serctl utility. It shows
# serctl start
Starting SER : 0(14877) read 3820494812 from /dev/urandom
0(14877) seeding PRNG with 656407867
0(14877) test random number 38195676
0(14877) parse error (1,1-2): parse error
It seems that my ser.cfg file has wrong format or other problem.
Any one has the same matther before?
Does that mean my ser.cfg file encounters "parse error" at line 1
and coloum 1 to 2?
This is a somewhat unique situation, but with my config i have some call
features that are hosted on another SIP aware application running on the
In a branch_route block i am checking to see if destination is myself to
avoid loops, however when i direct a message to, say "127.0.0.1:5090"
and OpenSER is configured to listen on port 5060 only the "if (uri ==
mysql)" statment matches even though the port is different.
It seems to me that this behaviour is not intuative given the use of the
statment, can anyone else comment on this?