I'm no BRI expert but it looks to me like your wanpipe is set up for
E1/EuroISDN. Where did you get this setup information?
-MC
It is autoconfigured by wancfg
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Is there a development version that I can check now?
On Mon, Mar 9, 2009, Anthony MinessaleAnthony Minessale wrote:
it's not released yet,
please wait for the announcement that it has been completed sometime in the
next week or 2.
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Hi all,
it seems there is no way to do this :(
It could be great to be able to:
- decide if RPID should be present or not in the B leg for an
outbound call
- make RPID header fully customizable with variables
- filter RPID for inbound call
I saw that kokoska rokoska created a jira
rod napsal(a):
Hi all,
it seems there is no way to do this :(
It could be great to be able to:
- decide if RPID should be present or not in the B leg for an
outbound call
- make RPID header fully customizable with variables
- filter RPID for inbound call
I saw that
Yes I know, it's deprecated but many peers still rely on this and
P-Asserted-ID is not widely spread (my own experience).
moreover if we could strip the RPID, we could write a new one, but It
could be very convenient to get access to the fields in this header for
manipulation.
thanks for
Hi All,
FS stopped working when NIC connection bounced.
Q1- Has anyone an explanation of what happened ?
Q2 - Is there a way to configure FS not to flip to another IP on a lost
network connection.
Q3 - Should an IP changed event be logged at a higher level e.g.
[CRITICAL] ?
NIC2
rod napsal(a):
... if we could strip the RPID, we could write a new one, but It
could be very convenient to get access to the fields in this header for
manipulation.
Yes, rod, this is exactly why I update bounty to $100 :-)
Thank you very much, rod, for support!
Best regards,
Latest SVN:
Send no extra caller id info:
{sip_cid_type=none}sofia/default/u...@example.com
Send RPID (default)
{sip_cid_type=rpid}sofia/default/u...@example.com
Send P-XXX-Identity
{sip_cid_type=pid}sofia/default/u...@example.com
Send RPID with chosen content
Anthony,
That is awesome. This is something that will be a BIG help.
SDR
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Anthony Minessale napsal(a):
Latest SVN:
Send no extra caller id info:
{sip_cid_type=none}sofia/default/u...@example.com mailto:u...@example.com
Send RPID (default)
{sip_cid_type=rpid}sofia/default/u...@example.com mailto:u...@example.com
Send P-XXX-Identity
The paypal button on the hompege will do ;)
On Tue, Mar 10, 2009 at 8:36 AM, kokoska rokoska kokoska.roko...@post.czwrote:
Anthony Minessale napsal(a):
Latest SVN:
Send no extra caller id info:
{sip_cid_type=none}sofia/default/u...@example.com mailto:
u...@example.com
Send RPID
conf/autoload_configs/sofia.conf.xml
global_settings
param name=log-level value=0/
!-- param name=auto-restart value=false/ --
param name=debug-presence value=0/
/global_settings
uncomment out the auto-restart.
On Tue, Mar 10, 2009 at 5:26 AM, Richard Lamkin
Hi Anthony,
thanks for this but I'd like to know if it's possible also to change
only the caller_id_name and caller_id_number without modifying the from
header.
ex: with the origination variables I get this
From: test sip:1...@172.29.0.5;tag=X4v4Kvt1B2DQF
Remote-Party-ID: test
ok if you are up to date you should be able to add
{sip_from_uri=sip:anonym...@anonymous.invalid} to your dial string.
On Tue, Mar 10, 2009 at 9:30 AM, rod kawa...@laposte.net wrote:
Hi Anthony,
thanks for this but I'd like to know if it's possible also to change
only the caller_id_name
Anthony Minessale napsal(a):
ok if you are up to date you should be able to add
{sip_from_uri=sip:anonym...@anonymous.invalid} to your dial string.
Many thanks, Anthony, for that feature!
It makes my life a lot easier :-)
Best regards,
kokoska.rokoska
Hello,
has anybody an idea?
regards
Helmut
On 09.03.2009 16:19, Helmut Kuper wrote:
Hello,
following scenario:
-Phone A is redirected unconditionally to phone C
-Phone B calls A
-Phone A replys with 302 and Dieversion header
-FS detects the 302 and sends out a new INVITE to C
I found
If you have nothing better to do drop by IRC
We are up to 193 users and about to cross 200 for the first time.
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
Hello,
are these variables only available at call setup time or can they be
changed during a call, e.g. before a call is being transferred to
another destination?
Best regards
Peter
Michael Collins schrieb:
On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
These should be available any time you are going to process a call thru the
dialplan and call a bridge on the call
From: Peter P GMX prometheus...@gmx.net
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 10 Mar 2009 23:20:39 +0100
To: freeswitch-users@lists.freeswitch.org
Subject:
Hey, I just implemented something like this and commited it to my contrib
directory (scripts/contrib/swk ) its a mixture of amf-php, ESL, and Flex...
Its not complete by anymeans and you need Flex3 to compile the UI...
Anyone wanting to throw some patches at it for other functionality are
welcome
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