Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B Ipbx
Asterisk Server 1.6.x for connect a lot
2012/4/25 Olivier CALVANO o.calv...@gmail.com
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B
Hi
i have a lot of error in the CLI of one of my Asterisk:
[Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit
of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation
not permitted
[Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit
of 0x8639de8
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :
2012/4/25 Olivier CALVANO
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
Thanks
Bryant
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Hi,
I have read the excellent information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
and believe I have an understanding of what is offered. I have a
couple of questions:
- Is it possible to update COLP/COLR when a SIP redirect occurs, or
when a SIP
Hi Carlos
It could help if you can get a trace of the call transfer from Nortel to SIP
extension on the Asterisk (1303), if no way to get from Nortel get from
Asterisk.
I guest operator try to make a bind call transfer, without wait complete DR2
signalling exchange at least minimal time
I have read the excellent information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
and believe I have an understanding of what is offered. I have a
couple of questions:
- Is it possible to update COLP/COLR when a SIP redirect occurs, or
when a
On 25 April 2012 16:55, Richard Mudgett rmudg...@digium.com wrote:
[snip]
- Is it possible to have the COLP/COLR information updated when a SIP
attended transfer is completed? If so how?
Transfers generate connected line update events automatically. The connected
line interception macros
On 04/25/2012 11:54 AM, Steve Davies wrote:
A further question... It appears that for SIP endpoints, this facility
only updates RPID and PAI headers? I have found that there appear to
be 4 different SIP CID-update mechanisms out there as follows:
- Update RPID and PAI (ITSP and trunks often
Daniel wrote:
Hi Group,
is in MeetMe any option to identify the own number (from the view of a
caller)?
I would like to write an option to set on the telephone an request for voice,
if the room is muted. That request should display on our Conference Control
Website and an Admin
List users,
I have an AudioCodes nCite 1000 SBC that is end-of-life and I'm
looking to replace it with open source software. I believe one of the
SIP proxy projects will fit my needs, but I'm a bit overwhelmed by
the number of choices and I'd like the advice of experienced users
before I venture
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what
version of
Kevin
I am using 1.8.x 10.x
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
Hi,
Is it possible yet to restart a single Dahdi span in any version of
Asterisk? (instead of all of them)
Thanks.
-- James
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New to Asterisk? Join us for
On 04/25/2012 04:45 PM, brya...@zktech.com wrote:
Kevin
I am using 1.8.x 10.x
Then you have SIP_CAUSE available, although you'll have to enable it
because it is off by default due to performance concerns.
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday, April 25, 2012 6:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code
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