Hi,
Following the start of the work for OpenSIPS 2.0, the priorities and
policies on releases changed.
Right now, most of development effort will be invested in the version
2.0 code. What this means:
A) Old (existing) design
- there will be no more major release for this design - that's it,
Hello Anca,
An ven., févr 12, 2010, opensipsl...@encambio.com schrieb:
>An ven., févr 12, 2010, Anca Vamanu schrieb:
>>It is ok to have cseq 2 on the other side if the received Invite
>>has cseq 1. The implementation uses cseq +1 on the other side :).
>>It should not be an issue.
>>
>You're right
Hi Daniel,
actually the alias_db module (as C code) supports multiple users with
the same alias. see:
http://www.opensips.org/html/docs/modules/1.6.x/alias_db.html#id228163
So, if you remove the DB constraint, it should work.
Regards,
Bogdan
Daniel Goepp wrote:
> Thanks, it helps...not
Hi Alexander,
To read digits after the call started, you need an IVR (media server) -
OpenSIPS cannot do something like that as it has no media support.
As I remember, Asterisk has a Read function for that -
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Regards,
Bogdan
Кузьмицкий Алекс
Hi,
Yes, that is the only way - if REGISTRAR module builds the reply, it
will do it in RFC way (adding all the registered contacts) and you
cannot change this behaviour.
Regards,
Bogdan
mayamatakeshi wrote:
>
> On Tue, Feb 16, 2010 at 12:54 AM, mayamatakeshi
> mailto:mayamatake...@gmail.com>>
Hi Jeff,
Now, according to the new release policy, the backport of this change
from trunk to 1.6 is doable :) I will take care or it
Regards,
Bogdan
Jeff Pyle wrote:
> Ha! I crack myself up. May I request this?
>
>
> - Jeff
>
>
> On Feb 15, 2010, at 8:37 AM, Bogdan-Andrei Iancu wrote:
>
>
Hi,
The received param is automatically added by OpenSIPS (as per RFC) as it
detects a difference between the IP address in the VIA hdr (the UAC's
IP) and the source IP at network level (the LB's IP).
There is no way to disable this (as it is standard SIP processing), but
if something really
Can you show litle example about using branches in this case?
I want to all calls(branches) starts dialing in parallel mode/
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Thanks for the offer, Bogdan. I've patched it up myself just to see that it
works as expected. In msg_translator.c:received_builder() I added the #ifdef
... #else ... #endif:
#ifdef NOTDEFINED
memcpy(buf, RECEIVED, RECEIVED_LEN);
if ( (tmp=ip_addr2a(source_ip))==0)
Hello,
I make use of uac_replace_from() in my scripts. I believe that is where this
error message is coming from:
ERROR:uac:restore_uri: new URI shorter than old URI
Can anyone explain what this really means? Having a new URI shorter than an
old one really doesn't seem like an error conditi
I`m trying next routing logic:
for alias 1009 i have two subscribers 1000 and 1001
When I try dialing from 1002 to 1009 ring only one phone(first element from
table dbalias).
Config blocks:
..
modparam("alias_db", "append_branches", 1)
..
alias_db_lookup("dbaliases");
11 matches
Mail list logo